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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 <string.h>
#include <sys/cdefs.h>
#include <sys/types.h>
#include "audio-base-utils.h"
#include "audio-base.h"
#include "audio-hal-enums.h"
#include "audio_common-base.h"
/*
* Annotation to tell clang that we intend to fall through from one case to
* another in a switch. Sourced from android-base/macros.h.
*/
#ifndef FALLTHROUGH_INTENDED
#ifdef __cplusplus
#define FALLTHROUGH_INTENDED [[fallthrough]]
#elif __has_attribute(fallthrough)
#define FALLTHROUGH_INTENDED __attribute__((__fallthrough__))
#else
#define FALLTHROUGH_INTENDED
#endif // __cplusplus
#endif // FALLTHROUGH_INTENDED
__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;
/* Null values for handles. */
enum {
AUDIO_IO_HANDLE_NONE = 0,
AUDIO_MODULE_HANDLE_NONE = 0,
AUDIO_PORT_HANDLE_NONE = 0,
AUDIO_PATCH_HANDLE_NONE = 0,
};
typedef enum {
#ifndef AUDIO_NO_SYSTEM_DECLARATIONS
AUDIO_MODE_INVALID = -2, // (-2)
AUDIO_MODE_CURRENT = -1, // (-1)
#endif // AUDIO_NO_SYSTEM_DECLARATIONS
AUDIO_MODE_NORMAL = HAL_AUDIO_MODE_NORMAL,
AUDIO_MODE_RINGTONE = HAL_AUDIO_MODE_RINGTONE,
AUDIO_MODE_IN_CALL = HAL_AUDIO_MODE_IN_CALL,
AUDIO_MODE_IN_COMMUNICATION = HAL_AUDIO_MODE_IN_COMMUNICATION,
AUDIO_MODE_CALL_SCREEN = HAL_AUDIO_MODE_CALL_SCREEN,
#ifndef AUDIO_NO_SYSTEM_DECLARATIONS
AUDIO_MODE_MAX = AUDIO_MODE_CALL_SCREEN,
AUDIO_MODE_CNT = AUDIO_MODE_MAX + 1,
#endif // AUDIO_NO_SYSTEM_DECLARATIONS
} audio_mode_t;
/* Do not change these values without updating their counterparts
* in frameworks/base/media/java/android/media/AudioAttributes.java
*/
typedef 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_FLAG_NO_MEDIA_PROJECTION = 0X400,
AUDIO_FLAG_MUTE_HAPTIC = 0x800,
AUDIO_FLAG_NO_SYSTEM_CAPTURE = 0X1000,
AUDIO_FLAG_CAPTURE_PRIVATE = 0X2000,
AUDIO_FLAG_CONTENT_SPATIALIZED = 0X4000,
AUDIO_FLAG_NEVER_SPATIALIZE = 0X8000,
} audio_flags_mask_t;
/* 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;
static const audio_attributes_t AUDIO_ATTRIBUTES_INITIALIZER = {
/* .content_type = */ AUDIO_CONTENT_TYPE_UNKNOWN,
/* .usage = */ AUDIO_USAGE_UNKNOWN,
/* .source = */ AUDIO_SOURCE_DEFAULT,
/* .flags = */ AUDIO_FLAG_NONE,
/* .tags = */ ""
};
static inline audio_attributes_t attributes_initializer(audio_usage_t usage)
{
audio_attributes_t attributes = AUDIO_ATTRIBUTES_INITIALIZER;
attributes.usage = usage;
return attributes;
}
static inline audio_attributes_t attributes_initializer_flags(audio_flags_mask_t flags)
{
audio_attributes_t attributes = AUDIO_ATTRIBUTES_INITIALIZER;
attributes.flags = flags;
return attributes;
}
static inline void audio_flags_to_audio_output_flags(
const audio_flags_mask_t audio_flags,
audio_output_flags_t *flags)
{
if ((audio_flags & AUDIO_FLAG_HW_AV_SYNC) != 0) {
*flags = (audio_output_flags_t)(*flags |
AUDIO_OUTPUT_FLAG_HW_AV_SYNC | AUDIO_OUTPUT_FLAG_DIRECT);
}
if ((audio_flags & AUDIO_FLAG_LOW_LATENCY) != 0) {
*flags = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_FAST);
}
// check deep buffer after flags have been modified above
if (*flags == AUDIO_OUTPUT_FLAG_NONE && (audio_flags & AUDIO_FLAG_DEEP_BUFFER) != 0) {
*flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
}
}
/* A unique ID allocated by AudioFlinger for use as an audio_io_handle_t, audio_session_t,
* audio_effect_handle_t, 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;
/* A unique ID with use AUDIO_UNIQUE_ID_USE_EFFECT */
typedef int audio_effect_handle_t;
/* Possible uses for an audio_unique_id_t */
typedef enum {
AUDIO_UNIQUE_ID_USE_UNSPECIFIED = 0,
AUDIO_UNIQUE_ID_USE_SESSION = 1, // audio_session_t
// for allocated sessions, not special AUDIO_SESSION_*
AUDIO_UNIQUE_ID_USE_MODULE = 2, // audio_module_handle_t
AUDIO_UNIQUE_ID_USE_EFFECT = 3, // audio_effect_handle_t
AUDIO_UNIQUE_ID_USE_PATCH = 4, // audio_patch_handle_t
AUDIO_UNIQUE_ID_USE_OUTPUT = 5, // audio_io_handle_t
AUDIO_UNIQUE_ID_USE_INPUT = 6, // audio_io_handle_t
AUDIO_UNIQUE_ID_USE_CLIENT = 7, // client-side players and recorders
// FIXME should move to a separate namespace;
// these IDs are allocated by AudioFlinger on client request,
// but are never used by AudioFlinger
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);
}
typedef enum {
AUDIO_SESSION_DEVICE = HAL_AUDIO_SESSION_DEVICE,
AUDIO_SESSION_OUTPUT_STAGE = HAL_AUDIO_SESSION_OUTPUT_STAGE,
AUDIO_SESSION_OUTPUT_MIX = HAL_AUDIO_SESSION_OUTPUT_MIX,
#ifndef AUDIO_NO_SYSTEM_DECLARATIONS
AUDIO_SESSION_ALLOCATE = 0,
AUDIO_SESSION_NONE = 0,
#endif
} audio_session_t;
/* Reserved audio_unique_id_t values. FIXME: not a complete list. */
#define AUDIO_UNIQUE_ID_ALLOCATE AUDIO_SESSION_ALLOCATE
/* returns true if the audio session ID corresponds to a global
* effect sessions (e.g. OUTPUT_MIX, OUTPUT_STAGE, or DEVICE).
*/
static inline bool audio_is_global_session(audio_session_t session) {
return session <= AUDIO_SESSION_OUTPUT_MIX;
}
/* These constants are used instead of "magic numbers" for
* channel counts.
*/
enum {
FCC_1 = 1,
FCC_2 = 2,
FCC_8 = 8,
FCC_12 = 12,
FCC_24 = 24,
FCC_26 = 26,
// FCC_LIMIT is the maximum PCM channel count supported through
// the mixing pipeline to the audio HAL.
//
// This can be adjusted onto a value such as FCC_12 or FCC_26
// if the device HAL can support it. Do not reduce below FCC_8.
FCC_LIMIT = FCC_12,
};
/* 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.
*/
/* 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 enum {
AUDIO_CHANNEL_REPRESENTATION_POSITION = 0x0u,
AUDIO_CHANNEL_REPRESENTATION_INDEX = 0x2u,
} 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));
}
#ifdef __cplusplus
// Some effects use `int32_t` directly for channel mask.
static inline uint32_t audio_channel_mask_get_representation(int32_t mask) {
return audio_channel_mask_get_representation(static_cast<audio_channel_mask_t>(mask));
}
#endif
/* 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;
/* 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;
audio_encapsulation_mode_t encapsulation_mode; // version 0.2:
int32_t content_id; // version 0.2: content id from tuner hal (0 if none)
int32_t sync_id; // version 0.2: sync id from tuner hal (0 if none)
} __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_2 AUDIO_MAKE_OFFLOAD_INFO_VERSION(0, 2)
#define AUDIO_OFFLOAD_INFO_VERSION_CURRENT AUDIO_OFFLOAD_INFO_VERSION_0_2
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 = */ AUDIO_CHANNEL_NONE,
/* .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,
/* .encapsulation_mode = */ AUDIO_ENCAPSULATION_MODE_NONE,
/* .content_id = */ 0,
/* .sync_id = */ 0,
};
/* 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 = */ AUDIO_CHANNEL_NONE,
/* .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,
/* .encapsulation_mode = */ AUDIO_ENCAPSULATION_MODE_NONE,
/* .content_id = */ 0,
/* .sync_id = */ 0,
},
/* .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
};
static inline audio_config_t audio_config_initializer(const audio_config_base_t *base)
{
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = base->sample_rate;
config.channel_mask = base->channel_mask;
config.format = base->format;
return config;
}
/* 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_configuration.xml file. The audio HAL
* will then implement gain control functions that will use the following data
* structures. */
/* 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 __builtin_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
/* a union to store port configuration flags. Declared as a type so can be reused
in framework code */
union audio_io_flags {
audio_input_flags_t input;
audio_output_flags_t output;
};
/* 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 */
};
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 = 0x1u,
AUDIO_PORT_CONFIG_CHANNEL_MASK = 0x2u,
AUDIO_PORT_CONFIG_FORMAT = 0x4u,
AUDIO_PORT_CONFIG_GAIN = 0x8u,
#ifndef AUDIO_NO_SYSTEM_DECLARATIONS
AUDIO_PORT_CONFIG_FLAGS = 0x10u,
#endif
AUDIO_PORT_CONFIG_ALL = AUDIO_PORT_CONFIG_SAMPLE_RATE |
AUDIO_PORT_CONFIG_CHANNEL_MASK |
AUDIO_PORT_CONFIG_FORMAT |
AUDIO_PORT_CONFIG_GAIN,
};
typedef enum {
AUDIO_LATENCY_LOW = 0,
AUDIO_LATENCY_NORMAL = 1,
} audio_mix_latency_class_t;
/* 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 */
#ifndef AUDIO_NO_SYSTEM_DECLARATIONS
union audio_io_flags flags; /* framework only: HW_AV_SYNC, DIRECT, ... */
#endif
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 audio profiles in audio port. The audio profiles are used in
* `struct audio_port_v7`. When converting between `struct audio_port` and
* `struct audio_port_v7`, the number of audio profiles in `struct audio_port_v7`
* must be the same as the number of formats in `struct audio_port`. Therefore,
* the maximum number of audio profiles must be the same as the maximum number
* of formats. */
#define AUDIO_PORT_MAX_AUDIO_PROFILES AUDIO_PORT_MAX_FORMATS
/* max number of extra audio descriptors in audio port. */
#define AUDIO_PORT_MAX_EXTRA_AUDIO_DESCRIPTORS AUDIO_PORT_MAX_FORMATS
/* max number of gain controls in audio port */
#define AUDIO_PORT_MAX_GAINS 16
/* max bytes of extra audio descriptor */
#define EXTRA_AUDIO_DESCRIPTOR_SIZE 32
/* 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];
#ifndef AUDIO_NO_SYSTEM_DECLARATIONS
uint32_t encapsulation_modes;
uint32_t encapsulation_metadata_types;
#endif
};
/* 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;
};
typedef enum {
AUDIO_STANDARD_NONE = 0,
AUDIO_STANDARD_EDID = 1,
} audio_standard_t;
/**
* Configuration described by hardware descriptor for a format that is unrecognized
* by the platform.
*/
struct audio_extra_audio_descriptor {
audio_standard_t standard;
unsigned int descriptor_length;
uint8_t descriptor[EXTRA_AUDIO_DESCRIPTOR_SIZE];
audio_encapsulation_type_t encapsulation_type;
};
/* configurations supported for a certain format */
struct audio_profile {
audio_format_t format;
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];
audio_encapsulation_type_t encapsulation_type;
};
struct audio_port_v7 {
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_audio_profiles; /* number of audio profiles in the following
array */
struct audio_profile audio_profiles[AUDIO_PORT_MAX_AUDIO_PROFILES];
unsigned int num_extra_audio_descriptors; /* number of extra audio descriptors in
the following array */
struct audio_extra_audio_descriptor
extra_audio_descriptors[AUDIO_PORT_MAX_EXTRA_AUDIO_DESCRIPTORS];
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;
};
/* Return true when a given uint8_t array is a valid short audio descriptor. This function just
* does basic validation by checking if the first value is not zero.
*/
static inline bool audio_is_valid_short_audio_descriptor(const uint8_t *shortAudioDescriptor,
size_t length) {
return length != 0 && *shortAudioDescriptor != 0;
}
static inline void audio_populate_audio_port_v7(
const struct audio_port *port, struct audio_port_v7 *portV7) {
portV7->id = port->id;
portV7->role = port->role;
portV7->type = port->type;
strncpy(portV7->name, port->name, AUDIO_PORT_MAX_NAME_LEN);
portV7->name[AUDIO_PORT_MAX_NAME_LEN-1] = '\0';
portV7->num_audio_profiles =
port->num_formats > AUDIO_PORT_MAX_AUDIO_PROFILES ?
AUDIO_PORT_MAX_AUDIO_PROFILES : port->num_formats;
for (size_t i = 0; i < portV7->num_audio_profiles; ++i) {
portV7->audio_profiles[i].format = port->formats[i];
portV7->audio_profiles[i].num_sample_rates = port->num_sample_rates;
memcpy(portV7->audio_profiles[i].sample_rates, port->sample_rates,
port->num_sample_rates * sizeof(unsigned int));
portV7->audio_profiles[i].num_channel_masks = port->num_channel_masks;
memcpy(portV7->audio_profiles[i].channel_masks, port->channel_masks,
port->num_channel_masks * sizeof(audio_channel_mask_t));
}
portV7->num_gains = port->num_gains;
memcpy(portV7->gains, port->gains, port->num_gains * sizeof(struct audio_gain));
memcpy(&portV7->active_config, &port->active_config, sizeof(struct audio_port_config));
memcpy(&portV7->ext, &port->ext, sizeof(port->ext));
}
/* Populate the data in `struct audio_port` using data from `struct audio_port_v7`. As the
* `struct audio_port_v7` use audio profiles to describe its capabilities, it may contain more
* data for sample rates or channel masks than the data that can be held by `struct audio_port`.
* Return true if all the data from `struct audio_port_v7` are converted to `struct audio_port`.
* Otherwise, return false.
*/
static inline bool audio_populate_audio_port(
const struct audio_port_v7 *portV7, struct audio_port *port) {
bool allDataConverted = true;
port->id = portV7->id;
port->role = portV7->role;
port->type = portV7->type;
strncpy(port->name, portV7->name, AUDIO_PORT_MAX_NAME_LEN);
port->name[AUDIO_PORT_MAX_NAME_LEN-1] = '\0';
port->num_formats =
portV7->num_audio_profiles > AUDIO_PORT_MAX_FORMATS ?
AUDIO_PORT_MAX_FORMATS : portV7->num_audio_profiles;
port->num_sample_rates = 0;
port->num_channel_masks = 0;
for (size_t i = 0; i < port->num_formats; ++i) {
port->formats[i] = portV7->audio_profiles[i].format;
for (size_t j = 0; j < portV7->audio_profiles[i].num_sample_rates; ++j) {
size_t k = 0;
for (; k < port->num_sample_rates; ++k) {
if (port->sample_rates[k] == portV7->audio_profiles[i].sample_rates[j]) {
break;
}
}
if (k == port->num_sample_rates) {
if (port->num_sample_rates >= AUDIO_PORT_MAX_SAMPLING_RATES) {
allDataConverted = false;
break;
}
port->sample_rates[port->num_sample_rates++] =
portV7->audio_profiles[i].sample_rates[j];
}
}
for (size_t j = 0; j < portV7->audio_profiles[i].num_channel_masks; ++j) {
size_t k = 0;
for (; k < port->num_channel_masks; ++k) {
if (port->channel_masks[k] == portV7->audio_profiles[i].channel_masks[j]) {
break;
}
}
if (k == port->num_channel_masks) {
if (port->num_channel_masks >= AUDIO_PORT_MAX_CHANNEL_MASKS) {
allDataConverted = false;
break;
}
port->channel_masks[port->num_channel_masks++] =
portV7->audio_profiles[i].channel_masks[j];
}
}
}
port->num_gains = portV7->num_gains;
memcpy(port->gains, portV7->gains, port->num_gains * sizeof(struct audio_gain));
memcpy(&port->active_config, &portV7->active_config, sizeof(struct audio_port_config));
memcpy(&port->ext, &portV7->ext, sizeof(port->ext));
return allDataConverted && portV7->num_extra_audio_descriptors == 0;
}
static inline bool audio_gain_config_are_equal(
const struct audio_gain_config *lhs, const struct audio_gain_config *rhs) {
if (lhs->mode != rhs->mode) return false;
if (lhs->mode & AUDIO_GAIN_MODE_JOINT) {
if (lhs->values[0] != rhs->values[0]) return false;
}
if (lhs->mode & (AUDIO_GAIN_MODE_CHANNELS | AUDIO_GAIN_MODE_RAMP)) {
if (lhs->channel_mask != rhs->channel_mask) return false;
for (int i = 0; i < __builtin_popcount(lhs->channel_mask); ++i) {
if (lhs->values[i] != rhs->values[i]) return false;
}
}
return lhs->ramp_duration_ms == rhs->ramp_duration_ms;
}
static inline bool audio_has_input_direction(audio_port_type_t type, audio_port_role_t role) {
switch (type) {
case AUDIO_PORT_TYPE_DEVICE:
switch (role) {
case AUDIO_PORT_ROLE_SOURCE: return true;
case AUDIO_PORT_ROLE_SINK: return false;
default: return false;
}
case AUDIO_PORT_TYPE_MIX:
switch (role) {
case AUDIO_PORT_ROLE_SOURCE: return false;
case AUDIO_PORT_ROLE_SINK: return true;
default: return false;
}
default: return false;
}
}
static inline bool audio_port_config_has_input_direction(const struct audio_port_config *port_cfg) {
return audio_has_input_direction(port_cfg->type, port_cfg->role);
}
static inline bool audio_port_configs_are_equal(
const struct audio_port_config *lhs, const struct audio_port_config *rhs) {
if (lhs->role != rhs->role || lhs->type != rhs->type) return false;
switch (lhs->type) {
case AUDIO_PORT_TYPE_NONE: break;
case AUDIO_PORT_TYPE_DEVICE:
if (lhs->ext.device.hw_module != rhs->ext.device.hw_module ||
lhs->ext.device.type != rhs->ext.device.type ||
strncmp(lhs->ext.device.address, rhs->ext.device.address,
AUDIO_DEVICE_MAX_ADDRESS_LEN) != 0) {
return false;
}
break;
case AUDIO_PORT_TYPE_MIX:
if (lhs->ext.mix.hw_module != rhs->ext.mix.hw_module ||
lhs->ext.mix.handle != rhs->ext.mix.handle) return false;
if (lhs->role == AUDIO_PORT_ROLE_SOURCE &&
lhs->ext.mix.usecase.stream != rhs->ext.mix.usecase.stream) return false;
else if (lhs->role == AUDIO_PORT_ROLE_SINK &&
lhs->ext.mix.usecase.source != rhs->ext.mix.usecase.source) return false;
break;
case AUDIO_PORT_TYPE_SESSION:
if (lhs->ext.session.session != rhs->ext.session.session) return false;
break;
default: return false;
}
return
lhs->config_mask == rhs->config_mask &&
#ifndef AUDIO_NO_SYSTEM_DECLARATIONS
((lhs->config_mask & AUDIO_PORT_CONFIG_FLAGS) == 0 ||
(audio_port_config_has_input_direction(lhs) ?
lhs->flags.input == rhs->flags.input :
lhs->flags.output == rhs->flags.output)) &&
#endif
((lhs->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) == 0 ||
lhs->sample_rate == rhs->sample_rate) &&
((lhs->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) == 0 ||
lhs->channel_mask == rhs->channel_mask) &&
((lhs->config_mask & AUDIO_PORT_CONFIG_FORMAT) == 0 ||
lhs->format == rhs->format) &&
((lhs->config_mask & AUDIO_PORT_CONFIG_GAIN) == 0 ||
audio_gain_config_are_equal(&lhs->gain, &rhs->gain));
}
static inline bool audio_gains_are_equal(const struct audio_gain* lhs, const struct audio_gain* rhs) {
return lhs->mode == rhs->mode &&
((lhs->mode & AUDIO_GAIN_MODE_CHANNELS) != AUDIO_GAIN_MODE_CHANNELS ||
lhs->channel_mask == rhs->channel_mask) &&
lhs->min_value == rhs->min_value &&
lhs->max_value == rhs->max_value &&
lhs->default_value == rhs->default_value &&
lhs->step_value == rhs->step_value &&
lhs->min_ramp_ms == rhs->min_ramp_ms &&
lhs->max_ramp_ms == rhs->max_ramp_ms;
}
// Define the helper functions of compare two audio_port/audio_port_v7 only in
// C++ as it is easier to compare the device capabilities.
#ifdef __cplusplus
extern "C++" {
#include <map>
#include <set>
#include <type_traits>
#include <utility>
#include <vector>
namespace {
static inline bool audio_gain_array_contains_all_elements_from(
const struct audio_gain gains[], const size_t numGains,
const struct audio_gain from[], size_t numFromGains) {
for (size_t i = 0; i < numFromGains; ++i) {
size_t j = 0;
for (;j < numGains; ++j) {
if (audio_gains_are_equal(&from[i], &gains[j])) {
break;
}
}
if (j == numGains) {
return false;
}
}
return true;
}
template <typename T, std::enable_if_t<std::is_same<T, struct audio_port>::value
|| std::is_same<T, struct audio_port_v7>::value, int> = 0>
static inline bool audio_ports_base_are_equal(const T* lhs, const T* rhs) {
if (lhs->id != rhs->id || lhs->role != rhs->role || lhs->type != rhs->type ||
strncmp(lhs->name, rhs->name, AUDIO_PORT_MAX_NAME_LEN) != 0 ||
lhs->num_gains != rhs->num_gains) {
return false;
}
switch (lhs->type) {
case AUDIO_PORT_TYPE_NONE: break;
case AUDIO_PORT_TYPE_DEVICE:
if (
#ifndef AUDIO_NO_SYSTEM_DECLARATIONS
lhs->ext.device.encapsulation_modes != rhs->ext.device.encapsulation_modes ||
lhs->ext.device.encapsulation_metadata_types !=
rhs->ext.device.encapsulation_metadata_types ||
#endif
lhs->ext.device.hw_module != rhs->ext.device.hw_module ||
lhs->ext.device.type != rhs->ext.device.type ||
strncmp(lhs->ext.device.address, rhs->ext.device.address,
AUDIO_DEVICE_MAX_ADDRESS_LEN) != 0) {
return false;
}
break;
case AUDIO_PORT_TYPE_MIX:
if (lhs->ext.mix.hw_module != rhs->ext.mix.hw_module ||
lhs->ext.mix.handle != rhs->ext.mix.handle ||
lhs->ext.mix.latency_class != rhs->ext.mix.latency_class) {
return false;
}
break;
case AUDIO_PORT_TYPE_SESSION:
if (lhs->ext.session.session != rhs->ext.session.session) {
return false;
}
break;
default:
return false;
}
if (!audio_gain_array_contains_all_elements_from(
lhs->gains, lhs->num_gains, rhs->gains, rhs->num_gains) ||
!audio_gain_array_contains_all_elements_from(
rhs->gains, rhs->num_gains, lhs->gains, lhs->num_gains)) {
return false;
}
return audio_port_configs_are_equal(&lhs->active_config, &rhs->active_config);
}
template <typename T, std::enable_if_t<std::is_same<T, audio_format_t>::value
|| std::is_same<T, unsigned int>::value
|| std::is_same<T, audio_channel_mask_t>::value, int> = 0>
static inline bool audio_capability_arrays_are_equal(
const T lhs[], unsigned int lsize, const T rhs[], unsigned int rsize) {
std::set<T> lhsSet(lhs, lhs + lsize);
std::set<T> rhsSet(rhs, rhs + rsize);
return lhsSet == rhsSet;
}
using AudioProfileMap =
std::map<audio_format_t,
std::pair<std::set<unsigned int>, std::set<audio_channel_mask_t>>>;
static inline AudioProfileMap getAudioProfileMap(
const struct audio_profile profiles[], unsigned int size) {
AudioProfileMap audioProfiles;
for (size_t i = 0; i < size; ++i) {
std::set<unsigned int> sampleRates(
profiles[i].sample_rates, profiles[i].sample_rates + profiles[i].num_sample_rates);
std::set<audio_channel_mask_t> channelMasks(
profiles[i].channel_masks,
profiles[i].channel_masks + profiles[i].num_channel_masks);
audioProfiles.emplace(profiles[i].format, std::make_pair(sampleRates, channelMasks));
}
return audioProfiles;
}
static inline bool audio_profile_arrays_are_equal(
const struct audio_profile lhs[], unsigned int lsize,
const struct audio_profile rhs[], unsigned int rsize) {
return getAudioProfileMap(lhs, lsize) == getAudioProfileMap(rhs, rsize);
}
using ExtraAudioDescriptorMap =std::map<audio_standard_t,
std::map<audio_encapsulation_type_t,
std::set<std::vector<uint8_t>>>>;
static inline ExtraAudioDescriptorMap getExtraAudioDescriptorMap(
const struct audio_extra_audio_descriptor extraAudioDescriptors[],
unsigned int numExtraAudioDescriptors) {
ExtraAudioDescriptorMap extraAudioDescriptorMap;
for (unsigned int i = 0; i < numExtraAudioDescriptors; ++i) {
extraAudioDescriptorMap[extraAudioDescriptors[i].standard]
[extraAudioDescriptors[i].encapsulation_type].insert(
std::vector<uint8_t>(
extraAudioDescriptors[i].descriptor,
extraAudioDescriptors[i].descriptor
+ extraAudioDescriptors[i].descriptor_length));
}
return extraAudioDescriptorMap;
}
static inline bool audio_extra_audio_descriptor_are_equal(
const struct audio_extra_audio_descriptor lhs[], unsigned int lsize,
const struct audio_extra_audio_descriptor rhs[], unsigned int rsize) {
return getExtraAudioDescriptorMap(lhs, lsize) == getExtraAudioDescriptorMap(rhs, rsize);
}
} // namespace
static inline bool audio_ports_are_equal(
const struct audio_port* lhs, const struct audio_port* rhs) {
if (!audio_ports_base_are_equal(lhs, rhs)) {
return false;
}
return audio_capability_arrays_are_equal(
lhs->formats, lhs->num_formats, rhs->formats, rhs->num_formats) &&
audio_capability_arrays_are_equal(
lhs->sample_rates, lhs->num_sample_rates,
rhs->sample_rates, rhs->num_sample_rates) &&
audio_capability_arrays_are_equal(
lhs->channel_masks, lhs->num_channel_masks,
rhs->channel_masks, rhs->num_channel_masks);
}
static inline bool audio_ports_v7_are_equal(
const struct audio_port_v7* lhs, const struct audio_port_v7* rhs) {
if (!audio_ports_base_are_equal(lhs, rhs)) {
return false;
}
return audio_profile_arrays_are_equal(
lhs->audio_profiles, lhs->num_audio_profiles,
rhs->audio_profiles, rhs->num_audio_profiles) &&
audio_extra_audio_descriptor_are_equal(
lhs->extra_audio_descriptors, lhs->num_extra_audio_descriptors,
rhs->extra_audio_descriptors, rhs->num_extra_audio_descriptors);
}
} // extern "C++"
#endif // __cplusplus
/* 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
/** @TODO export from .hal */
typedef enum {
NONE = 0x0,
/**
* Only set this flag if applications can access the audio buffer memory
* shared with the backend (usually DSP) _without_ security issue.
*
* Setting this flag also implies that Binder will allow passing the shared memory FD
* to applications.
*
* That usually implies that the kernel will prevent any access to the
* memory surrounding the audio buffer as it could lead to a security breach.
*
* For example, a "/dev/snd/" file descriptor generally is not shareable,
* but an "anon_inode:dmabuffer" file descriptor is shareable.
* See also Linux kernel's dma_buf.
*
* This flag is required to support AAudio exclusive mode:
* See: https://source.android.com/devices/audio/aaudio
*/
AUDIO_MMAP_APPLICATION_SHAREABLE = 0x1,
} audio_mmap_buffer_flag;
/**
* 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 */
audio_mmap_buffer_flag flags; /**< Attributes describing the buffer. */
};
/**
* 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 playback 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 record track for an out stream. */
typedef struct record_track_metadata {
audio_source_t source;
float gain; // Normalized linear volume. 0=silence, 1=0dbfs...
// For record tracks originating from a software patch, the dest_device
// fields provide information about the downstream device.
audio_devices_t dest_device;
char dest_device_address[AUDIO_DEVICE_MAX_ADDRESS_LEN];
} record_track_metadata_t;
/** Metadata of a playback track for an in stream. */
typedef struct playback_track_metadata_v7 {
struct playback_track_metadata base;
audio_channel_mask_t channel_mask;
char tags[AUDIO_ATTRIBUTES_TAGS_MAX_SIZE]; /* UTF8 */
} playback_track_metadata_v7_t;
/** Metadata of a record track for an out stream. */
typedef struct record_track_metadata_v7 {
struct record_track_metadata base;
audio_channel_mask_t channel_mask;
char tags[AUDIO_ATTRIBUTES_TAGS_MAX_SIZE]; /* UTF8 */
} record_track_metadata_v7_t;
static inline void playback_track_metadata_to_v7(struct playback_track_metadata_v7 *dst,
const struct playback_track_metadata *src) {
dst->base = *src;
dst->channel_mask = AUDIO_CHANNEL_NONE;
dst->tags[0] = '\0';
}
static inline void playback_track_metadata_from_v7(struct playback_track_metadata *dst,
const struct playback_track_metadata_v7 *src) {
*dst = src->base;
}
static inline void record_track_metadata_to_v7(struct record_track_metadata_v7 *dst,
const struct record_track_metadata *src) {
dst->base = *src;
dst->channel_mask = AUDIO_CHANNEL_NONE;
dst->tags[0] = '\0';
}
static inline void record_track_metadata_from_v7(struct record_track_metadata *dst,
const struct record_track_metadata_v7 *src) {
*dst = src->base;
}
/******************************
* Helper functions
*****************************/
// see also: std::binary_search
// search range [left, right)
static inline bool audio_binary_search_device_array(const audio_devices_t audio_array[],
size_t left, size_t right,
audio_devices_t target)
{
if (right <= left || target < audio_array[left] || target > audio_array[right - 1]) {
return false;
}
while (left < right) {
const size_t mid = left + (right - left) / 2;
if (audio_array[mid] == target) {
return true;
} else if (audio_array[mid] < target) {
left = mid + 1;
} else {
right = mid;
}
}
return false;
}
static inline bool audio_is_output_device(audio_devices_t device)
{
switch (device) {
case AUDIO_DEVICE_OUT_SPEAKER_SAFE:
case AUDIO_DEVICE_OUT_SPEAKER:
case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
case AUDIO_DEVICE_OUT_WIRED_HEADSET:
case AUDIO_DEVICE_OUT_USB_HEADSET:
case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
case AUDIO_DEVICE_OUT_EARPIECE:
case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
case AUDIO_DEVICE_OUT_TELEPHONY_TX:
// Search the most common devices first as these devices are most likely
// to be used. Put the most common devices in the order of the likelihood
// of usage to get a quick return.
return true;
default:
// Binary seach all devices if the device is not a most common device.
return audio_binary_search_device_array(
AUDIO_DEVICE_OUT_ALL_ARRAY, 0 /*left*/, AUDIO_DEVICE_OUT_CNT, device);
}
}
static inline bool audio_is_input_device(audio_devices_t device)
{
switch (device) {
case AUDIO_DEVICE_IN_BUILTIN_MIC:
case AUDIO_DEVICE_IN_BACK_MIC:
case AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET:
case AUDIO_DEVICE_IN_WIRED_HEADSET:
case AUDIO_DEVICE_IN_USB_HEADSET:
case AUDIO_DEVICE_IN_REMOTE_SUBMIX:
case AUDIO_DEVICE_IN_TELEPHONY_RX:
// Search the most common devices first as these devices are most likely
// to be used. Put the most common devices in the order of the likelihood
// of usage to get a quick return.
return true;
default:
// Binary seach all devices if the device is not a most common device.
return audio_binary_search_device_array(
AUDIO_DEVICE_IN_ALL_ARRAY, 0 /*left*/, AUDIO_DEVICE_IN_CNT, device);
}
}
#ifdef __cplusplus
// Some effects use `uint32_t` directly for device.
static inline bool audio_is_input_device(uint32_t device) {
return audio_is_input_device(static_cast<audio_devices_t>(device));
}
// This needs to be used when `audio_is_input_device` is passed
// to an STL algorithm, as otherwise the compiler can't resolve
// the overload at that point--the type of the container elements
// doesn't appear in the predicate parameter type definition.
const auto audio_call_is_input_device = [](auto x) { return audio_is_input_device(x); };
#endif
// TODO: this function expects a combination of audio device types as parameter. It should
// be deprecated as audio device types should not be use as bit mask any more since R.
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)
{
return device == AUDIO_DEVICE_IN_BLUETOOTH_A2DP;
}
static inline bool audio_is_a2dp_out_device(audio_devices_t device)
{
return audio_binary_search_device_array(
AUDIO_DEVICE_OUT_ALL_A2DP_ARRAY, 0 /*left*/, AUDIO_DEVICE_OUT_A2DP_CNT, device);
}
// 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_out_sco_device(audio_devices_t device)
{
return audio_binary_search_device_array(
AUDIO_DEVICE_OUT_ALL_SCO_ARRAY, 0 /*left*/, AUDIO_DEVICE_OUT_SCO_CNT, device);
}
static inline bool audio_is_bluetooth_in_sco_device(audio_devices_t device)
{
return audio_binary_search_device_array(
AUDIO_DEVICE_IN_ALL_SCO_ARRAY, 0 /*left*/, AUDIO_DEVICE_IN_SCO_CNT, device);
}
static inline bool audio_is_bluetooth_sco_device(audio_devices_t device)
{
return audio_is_bluetooth_out_sco_device(device) ||
audio_is_bluetooth_in_sco_device(device);
}
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 audio_binary_search_device_array(
AUDIO_DEVICE_OUT_ALL_USB_ARRAY, 0 /*left*/, AUDIO_DEVICE_OUT_USB_CNT, device);
}
static inline bool audio_is_usb_in_device(audio_devices_t device)
{
return audio_binary_search_device_array(
AUDIO_DEVICE_IN_ALL_USB_ARRAY, 0 /*left*/, AUDIO_DEVICE_IN_USB_CNT, device);
}
/* 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)
{
return device == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ||
device == AUDIO_DEVICE_IN_REMOTE_SUBMIX;
}
static inline bool audio_is_digital_out_device(audio_devices_t device)
{
return audio_binary_search_device_array(
AUDIO_DEVICE_OUT_ALL_DIGITAL_ARRAY, 0 /*left*/, AUDIO_DEVICE_OUT_DIGITAL_CNT, device);
}
static inline bool audio_is_digital_in_device(audio_devices_t device)
{
return audio_binary_search_device_array(
AUDIO_DEVICE_IN_ALL_DIGITAL_ARRAY, 0 /*left*/, AUDIO_DEVICE_IN_DIGITAL_CNT, device);
}
static inline bool audio_device_is_digital(audio_devices_t device) {
return audio_is_digital_in_device(device) ||
audio_is_digital_out_device(device);
}
static inline bool audio_is_ble_out_device(audio_devices_t device)
{
return audio_binary_search_device_array(
AUDIO_DEVICE_OUT_ALL_BLE_ARRAY, 0 /*left*/, AUDIO_DEVICE_OUT_BLE_CNT, device);
}
static inline bool audio_is_ble_in_device(audio_devices_t device)
{
return audio_binary_search_device_array(
AUDIO_DEVICE_IN_ALL_BLE_ARRAY, 0 /*left*/, AUDIO_DEVICE_IN_BLE_CNT, device);
}
static inline bool audio_is_ble_device(audio_devices_t device) {
return audio_is_ble_in_device(device) ||
audio_is_ble_out_device(device);
}
/* 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;
}
FALLTHROUGH_INTENDED;
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;
}
FALLTHROUGH_INTENDED;
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;
FALLTHROUGH_INTENDED;
case AUDIO_CHANNEL_REPRESENTATION_INDEX:
return __builtin_popcount(bits);
default:
return 0;
}
}
#ifdef __cplusplus
// FIXME(b/169889714): buffer_config_t uses `uint32_t` for the mask.
// A lot of effects code thus use `uint32_t` directly.
static inline uint32_t audio_channel_count_from_in_mask(uint32_t mask) {
return audio_channel_count_from_in_mask(static_cast<audio_channel_mask_t>(mask));
}
#endif
/* 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;
FALLTHROUGH_INTENDED;
case AUDIO_CHANNEL_REPRESENTATION_INDEX:
return __builtin_popcount(bits);
default:
return 0;
}
}
#ifdef __cplusplus
// FIXME(b/169889714): buffer_config_t uses `uint32_t` for the mask.
// A lot of effects code thus use `uint32_t` directly.
static inline uint32_t audio_channel_count_from_out_mask(uint32_t mask) {
return audio_channel_count_from_out_mask(static_cast<audio_channel_mask_t>(mask));
}
#endif
/* 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: // 2.1
bits = AUDIO_CHANNEL_OUT_STEREO | AUDIO_CHANNEL_OUT_LOW_FREQUENCY;
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 FCC_8:
bits = AUDIO_CHANNEL_OUT_7POINT1;
break;
case FCC_12:
bits = AUDIO_CHANNEL_OUT_7POINT1POINT4;
break;
case FCC_24:
bits = AUDIO_CHANNEL_OUT_22POINT2;
break;
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;
default:
if (channel_count <= FCC_LIMIT) {
return audio_channel_mask_for_index_assignment_from_count(channel_count);
}
return AUDIO_CHANNEL_INVALID;
}
return audio_channel_mask_from_representation_and_bits(
AUDIO_CHANNEL_REPRESENTATION_POSITION, bits);
}
/* Derive a default haptic channel mask from a channel count.
*/
static inline audio_channel_mask_t haptic_channel_mask_from_count(uint32_t channel_count)
{
switch(channel_count) {
case 0:
return AUDIO_CHANNEL_NONE;
case 1:
return AUDIO_CHANNEL_OUT_HAPTIC_A;
case 2:
return AUDIO_CHANNEL_OUT_HAPTIC_AB;
default:
return AUDIO_CHANNEL_INVALID;
}
}
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 audio_channel_mask_t audio_channel_mask_out_to_in(audio_channel_mask_t out)
{
switch (out) {
case AUDIO_CHANNEL_OUT_MONO:
return AUDIO_CHANNEL_IN_MONO;
case AUDIO_CHANNEL_OUT_STEREO:
return AUDIO_CHANNEL_IN_STEREO;
case AUDIO_CHANNEL_OUT_5POINT1:
return AUDIO_CHANNEL_IN_5POINT1;
case AUDIO_CHANNEL_OUT_3POINT1POINT2:
return AUDIO_CHANNEL_IN_3POINT1POINT2;
case AUDIO_CHANNEL_OUT_3POINT0POINT2:
return AUDIO_CHANNEL_IN_3POINT0POINT2;
case AUDIO_CHANNEL_OUT_2POINT1POINT2:
return AUDIO_CHANNEL_IN_2POINT1POINT2;
case AUDIO_CHANNEL_OUT_2POINT0POINT2:
return AUDIO_CHANNEL_IN_2POINT0POINT2;
default:
return AUDIO_CHANNEL_INVALID;
}
}
static inline bool audio_channel_position_mask_is_out_canonical(audio_channel_mask_t channelMask)
{
if (audio_channel_mask_get_representation(channelMask)
!= AUDIO_CHANNEL_REPRESENTATION_POSITION) {
return false;
}
const uint32_t audioChannelCount = audio_channel_count_from_out_mask(
(audio_channel_mask_t)(channelMask & ~AUDIO_CHANNEL_HAPTIC_ALL));
const uint32_t hapticChannelCount = audio_channel_count_from_out_mask(
(audio_channel_mask_t)(channelMask & AUDIO_CHANNEL_HAPTIC_ALL));
return channelMask == (audio_channel_mask_t)(
audio_channel_out_mask_from_count(audioChannelCount) |
haptic_channel_mask_from_count(hapticChannelCount));
}
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:
return true;
case AUDIO_FORMAT_AAC:
switch (format) {
case AUDIO_FORMAT_AAC:
case AUDIO_FORMAT_AAC_MAIN:
case AUDIO_FORMAT_AAC_LC:
case AUDIO_FORMAT_AAC_SSR:
case AUDIO_FORMAT_AAC_LTP:
case AUDIO_FORMAT_AAC_HE_V1:
case AUDIO_FORMAT_AAC_SCALABLE:
case AUDIO_FORMAT_AAC_ERLC:
case AUDIO_FORMAT_AAC_LD:
case AUDIO_FORMAT_AAC_HE_V2:
case AUDIO_FORMAT_AAC_ELD:
case AUDIO_FORMAT_AAC_XHE:
return true;
default:
return false;
}
/* not reached */
case AUDIO_FORMAT_HE_AAC_V1:
case AUDIO_FORMAT_HE_AAC_V2:
case AUDIO_FORMAT_VORBIS:
case AUDIO_FORMAT_OPUS:
case AUDIO_FORMAT_AC3:
return true;
case AUDIO_FORMAT_E_AC3:
switch (format) {
case AUDIO_FORMAT_E_AC3:
case AUDIO_FORMAT_E_AC3_JOC:
return true;
default:
return false;
}
/* not reached */
case AUDIO_FORMAT_DTS:
case AUDIO_FORMAT_DTS_HD:
case AUDIO_FORMAT_IEC60958:
case AUDIO_FORMAT_IEC61937:
case AUDIO_FORMAT_DOLBY_TRUEHD:
case AUDIO_FORMAT_EVRC:
case AUDIO_FORMAT_EVRCB:
case AUDIO_FORMAT_EVRCWB:
case AUDIO_FORMAT_EVRCNW:
case AUDIO_FORMAT_AAC_ADIF:
case AUDIO_FORMAT_WMA:
case AUDIO_FORMAT_WMA_PRO:
case AUDIO_FORMAT_AMR_WB_PLUS:
case AUDIO_FORMAT_MP2:
case AUDIO_FORMAT_QCELP:
case AUDIO_FORMAT_DSD:
case AUDIO_FORMAT_FLAC:
case AUDIO_FORMAT_ALAC:
case AUDIO_FORMAT_APE:
return true;
case AUDIO_FORMAT_AAC_ADTS:
switch (format) {
case AUDIO_FORMAT_AAC_ADTS:
case AUDIO_FORMAT_AAC_ADTS_MAIN:
case AUDIO_FORMAT_AAC_ADTS_LC:
case AUDIO_FORMAT_AAC_ADTS_SSR:
case AUDIO_FORMAT_AAC_ADTS_LTP:
case AUDIO_FORMAT_AAC_ADTS_HE_V1:
case AUDIO_FORMAT_AAC_ADTS_SCALABLE:
case AUDIO_FORMAT_AAC_ADTS_ERLC:
case AUDIO_FORMAT_AAC_ADTS_LD:
case AUDIO_FORMAT_AAC_ADTS_HE_V2:
case AUDIO_FORMAT_AAC_ADTS_ELD:
case AUDIO_FORMAT_AAC_ADTS_XHE:
return true;
default:
return false;
}
/* not reached */
case AUDIO_FORMAT_SBC:
case AUDIO_FORMAT_APTX:
case AUDIO_FORMAT_APTX_HD:
case AUDIO_FORMAT_AC4:
case AUDIO_FORMAT_LDAC:
return true;
case AUDIO_FORMAT_MAT:
switch (format) {
case AUDIO_FORMAT_MAT:
case AUDIO_FORMAT_MAT_1_0:
case AUDIO_FORMAT_MAT_2_0:
case AUDIO_FORMAT_MAT_2_1:
return true;
default:
return false;
}
/* not reached */
case AUDIO_FORMAT_AAC_LATM:
switch (format) {
case AUDIO_FORMAT_AAC_LATM:
case AUDIO_FORMAT_AAC_LATM_LC:
case AUDIO_FORMAT_AAC_LATM_HE_V1:
case AUDIO_FORMAT_AAC_LATM_HE_V2:
return true;
default:
return false;
}
/* not reached */
case AUDIO_FORMAT_CELT:
case AUDIO_FORMAT_APTX_ADAPTIVE:
case AUDIO_FORMAT_LHDC:
case AUDIO_FORMAT_LHDC_LL:
case AUDIO_FORMAT_APTX_TWSP:
case AUDIO_FORMAT_LC3:
// case AUDIO_FORMAT_APTX_ADAPTIVE_QLEA:
return true;
case AUDIO_FORMAT_MPEGH:
switch (format) {
case AUDIO_FORMAT_MPEGH_BL_L3:
case AUDIO_FORMAT_MPEGH_BL_L4:
case AUDIO_FORMAT_MPEGH_LC_L3:
case AUDIO_FORMAT_MPEGH_LC_L4:
return true;
default:
return false;
}
/* not reached */
case AUDIO_FORMAT_DTS_UHD:
case AUDIO_FORMAT_DRA:
return true;
default:
return false;
}
}
static inline bool audio_is_iec61937_compatible(audio_format_t format)
{
switch (format) {
case AUDIO_FORMAT_AC3: // IEC 61937-3:2017
case AUDIO_FORMAT_AC4: // IEC 61937-14:2017
case AUDIO_FORMAT_E_AC3: // IEC 61937-3:2017
case AUDIO_FORMAT_E_AC3_JOC: // IEC 61937-3:2017
case AUDIO_FORMAT_MAT: // IEC 61937-9:2017
case AUDIO_FORMAT_MAT_1_0: // IEC 61937-9:2017
case AUDIO_FORMAT_MAT_2_0: // IEC 61937-9:2017
case AUDIO_FORMAT_MAT_2_1: // IEC 61937-9:2017
case AUDIO_FORMAT_MPEGH_BL_L3: // IEC 61937-13:2018
case AUDIO_FORMAT_MPEGH_BL_L4: // IEC 61937-13:2018
case AUDIO_FORMAT_MPEGH_LC_L3: // IEC 61937-13:2018
case AUDIO_FORMAT_MPEGH_LC_L4: // IEC 61937-13:2018
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_source_address=");
char param[kSize];
if (device == AUDIO_DEVICE_IN_BLUETOOTH_A2DP) {
snprintf(param, kSize, "%s=%s", "a2dp_source_address", address);
} else if (audio_is_a2dp_out_device(device)) {
snprintf(param, kSize, "%s=%s", "a2dp_sink_address", address);
} else if (audio_is_remote_submix_device(device)) {
snprintf(param, kSize, "%s=%s", "mix", address);
} else {
snprintf(param, kSize, "%s", address);
}
return strdup(param);
}
static inline bool audio_is_valid_audio_source(audio_source_t audioSource)
{
switch (audioSource) {
case AUDIO_SOURCE_MIC:
case AUDIO_SOURCE_VOICE_UPLINK:
case AUDIO_SOURCE_VOICE_DOWNLINK:
case AUDIO_SOURCE_VOICE_CALL:
case AUDIO_SOURCE_CAMCORDER:
case AUDIO_SOURCE_VOICE_RECOGNITION:
case AUDIO_SOURCE_VOICE_COMMUNICATION:
case AUDIO_SOURCE_REMOTE_SUBMIX:
case AUDIO_SOURCE_UNPROCESSED:
case AUDIO_SOURCE_VOICE_PERFORMANCE:
case AUDIO_SOURCE_ECHO_REFERENCE:
case AUDIO_SOURCE_FM_TUNER:
#ifndef AUDIO_NO_SYSTEM_DECLARATIONS
case AUDIO_SOURCE_HOTWORD:
#endif // AUDIO_NO_SYSTEM_DECLARATIONS
case AUDIO_SOURCE_ULTRASOUND:
return true;
default:
return false;
}
}
#ifndef AUDIO_NO_SYSTEM_DECLARATIONS
static inline bool audio_port_config_has_hw_av_sync(const struct audio_port_config *port_cfg) {
if (!(port_cfg->config_mask & AUDIO_PORT_CONFIG_FLAGS)) {
return false;
}
return audio_port_config_has_input_direction(port_cfg) ?
port_cfg->flags.input & AUDIO_INPUT_FLAG_HW_AV_SYNC
: port_cfg->flags.output & AUDIO_OUTPUT_FLAG_HW_AV_SYNC;
}
static inline bool audio_patch_has_hw_av_sync(const struct audio_patch *patch) {
for (unsigned int i = 0; i < patch->num_sources; ++i) {
if (audio_port_config_has_hw_av_sync(&patch->sources[i])) return true;
}
for (unsigned int i = 0; i < patch->num_sinks; ++i) {
if (audio_port_config_has_hw_av_sync(&patch->sinks[i])) return true;
}
return false;
}
static inline bool audio_patch_is_valid(const struct audio_patch *patch) {
// Note that patch can have no sinks.
return patch->num_sources != 0 && patch->num_sources <= AUDIO_PATCH_PORTS_MAX &&
patch->num_sinks <= AUDIO_PATCH_PORTS_MAX;
}
// Note that when checking for equality the order of ports must match.
// Patches will not be equivalent if they contain the same ports but they are permuted differently.
static inline bool audio_patches_are_equal(
const struct audio_patch *lhs, const struct audio_patch *rhs) {
if (!audio_patch_is_valid(lhs) || !audio_patch_is_valid(rhs)) return false;
if (lhs->num_sources != rhs->num_sources || lhs->num_sinks != rhs->num_sinks) return false;
for (unsigned int i = 0; i < lhs->num_sources; ++i) {
if (!audio_port_configs_are_equal(&lhs->sources[i], &rhs->sources[i])) return false;
}
for (unsigned int i = 0; i < lhs->num_sinks; ++i) {
if (!audio_port_configs_are_equal(&lhs->sinks[i], &rhs->sinks[i])) return false;
}
return true;
}
#endif
// 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;
/* 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;
/* 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;
};
typedef enum {
#ifndef AUDIO_NO_SYSTEM_DECLARATIONS
AUDIO_TIMESTRETCH_FALLBACK_CUT_REPEAT = -1, // (framework only) for speed <1.0 will truncate
// frames, for speed > 1.0 will repeat frames
AUDIO_TIMESTRETCH_FALLBACK_DEFAULT = 0, // (framework only) system determines behavior
#endif
/* Set all processed frames to zero. */
AUDIO_TIMESTRETCH_FALLBACK_MUTE = HAL_AUDIO_TIMESTRETCH_FALLBACK_MUTE,
/* Stop processing and indicate an error. */
AUDIO_TIMESTRETCH_FALLBACK_FAIL = HAL_AUDIO_TIMESTRETCH_FALLBACK_FAIL,
} audio_timestretch_fallback_mode_t;
// AUDIO_TIMESTRETCH_SPEED_MIN and AUDIO_TIMESTRETCH_SPEED_MAX define the min and max time stretch
// speeds supported by the system. These are enforced by the system and values outside this range
// will result in a runtime error.
// Depending on the AudioPlaybackRate::mStretchMode, the effective limits might be narrower than
// the ones specified here
// AUDIO_TIMESTRETCH_SPEED_MIN_DELTA is the minimum absolute speed difference that might trigger a
// parameter update
#define AUDIO_TIMESTRETCH_SPEED_MIN 0.01f
#define AUDIO_TIMESTRETCH_SPEED_MAX 20.0f
#define AUDIO_TIMESTRETCH_SPEED_NORMAL 1.0f
#define AUDIO_TIMESTRETCH_SPEED_MIN_DELTA 0.0001f
// AUDIO_TIMESTRETCH_PITCH_MIN and AUDIO_TIMESTRETCH_PITCH_MAX define the min and max time stretch
// pitch shifting supported by the system. These are not enforced by the system and values
// outside this range might result in a pitch different than the one requested.
// Depending on the AudioPlaybackRate::mStretchMode, the effective limits might be narrower than
// the ones specified here.
// AUDIO_TIMESTRETCH_PITCH_MIN_DELTA is the minimum absolute pitch difference that might trigger a
// parameter update
#define AUDIO_TIMESTRETCH_PITCH_MIN 0.25f
#define AUDIO_TIMESTRETCH_PITCH_MAX 4.0f
#define AUDIO_TIMESTRETCH_PITCH_NORMAL 1.0f
#define AUDIO_TIMESTRETCH_PITCH_MIN_DELTA 0.0001f
//Limits for AUDIO_TIMESTRETCH_STRETCH_VOICE mode
#define TIMESTRETCH_SONIC_SPEED_MIN 0.1f
#define TIMESTRETCH_SONIC_SPEED_MAX 6.0f
struct audio_playback_rate {
float mSpeed;
float mPitch;
audio_timestretch_stretch_mode_t mStretchMode;
audio_timestretch_fallback_mode_t mFallbackMode;
};
typedef struct audio_playback_rate audio_playback_rate_t;
static const audio_playback_rate_t AUDIO_PLAYBACK_RATE_INITIALIZER = {
/* .mSpeed = */ AUDIO_TIMESTRETCH_SPEED_NORMAL,
/* .mPitch = */ AUDIO_TIMESTRETCH_PITCH_NORMAL,
/* .mStretchMode = */ AUDIO_TIMESTRETCH_STRETCH_DEFAULT,
/* .mFallbackMode = */ AUDIO_TIMESTRETCH_FALLBACK_FAIL
};
#ifndef AUDIO_NO_SYSTEM_DECLARATIONS
typedef enum {
AUDIO_OFFLOAD_NOT_SUPPORTED = 0,
AUDIO_OFFLOAD_SUPPORTED = 1,
AUDIO_OFFLOAD_GAPLESS_SUPPORTED = 2
} audio_offload_mode_t;
#endif // AUDIO_NO_SYSTEM_DECLARATIONS
__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"
#define AUDIO_HARDWARE_MODULE_ID_MSD "msd"
/**
* 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"
/* User's preferred audio language setting (in ISO 639-2/T three-letter string code)
* used to select a specific language presentation for next generation audio codecs. */
#define AUDIO_PARAMETER_KEY_AUDIO_LANGUAGE_PREFERRED "audio_language_preferred"
/**
* 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"
/**
* For querying device supported encapsulation capabilities. All returned values are integer,
* which are bit fields composed from using encapsulation capability values as position bits.
* Encapsulation capability values are defined in audio_encapsulation_mode_t and
* audio_encapsulation_metadata_type_t. For instance, if the supported encapsulation mode is
* AUDIO_ENCAPSULATION_MODE_ELEMENTARY_STREAM, the returned value is
* "supEncapsulationModes=1 << AUDIO_ENCAPSULATION_MODE_ELEMENTARY_STREAM".
* When querying device supported encapsulation capabilities, the key should use device type
* and address so that it is able to identify the device. The device will be a key. The device
* type will be the value of key AUDIO_PARAMETER_STREAM_ROUTING.
*/
#define AUDIO_PARAMETER_DEVICE_SUP_ENCAPSULATION_MODES "supEncapsulationModes"
#define AUDIO_PARAMETER_DEVICE_SUP_ENCAPSULATION_METADATA_TYPES "supEncapsulationMetadataTypes"
/* Query additional delay in millisecond on each output device. */
#define AUDIO_PARAMETER_DEVICE_ADDITIONAL_OUTPUT_DELAY "additional_output_device_delay"
#define AUDIO_PARAMETER_DEVICE_MAX_ADDITIONAL_OUTPUT_DELAY "max_additional_output_device_delay"
/**
* 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