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LeafOS / LeafOS-Project / android_hardware_qcom_audio / bf1d47f32e0597a6950b2549b2c0978415ffb814 / . / hal / audio_hw.h
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/*
* Copyright (c) 2013-2020, The Linux Foundation. All rights reserved.
* Not a contribution.
*
* Copyright (C) 2013 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*
* This file was modified by DTS, Inc. The portions of the
* code modified by DTS, Inc are copyrighted and
* licensed separately, as follows:
*
* (C) 2014 DTS, Inc.
*
* 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 QCOM_AUDIO_HW_H
#define QCOM_AUDIO_HW_H
#include <stdlib.h>
#include <cutils/str_parms.h>
#include <cutils/list.h>
#include <cutils/hashmap.h>
#include <hardware/audio.h>
#include <tinyalsa/asoundlib.h>
#include <tinycompress/tinycompress.h>
#include <audio_route/audio_route.h>
#include <audio_utils/ErrorLog.h>
#include <audio_utils/Statistics.h>
#include <audio_utils/clock.h>
#include "audio_defs.h"
#include "voice.h"
#include "audio_hw_extn_api.h"
#include "device_utils.h"
#if LINUX_ENABLED
#if defined(__LP64__)
#define VISUALIZER_LIBRARY_PATH "/usr/lib64/libqcomvisualizer.so"
#define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "/usr/lib64/libqcompostprocbundle.so"
#define ADM_LIBRARY_PATH "/usr/lib64/libadm.so"
#else
#define VISUALIZER_LIBRARY_PATH "/usr/lib/libqcomvisualizer.so"
#define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "/usr/lib/libqcompostprocbundle.so"
#define ADM_LIBRARY_PATH "/usr/lib/libadm.so"
#endif
#else
#define VISUALIZER_LIBRARY_PATH "/vendor/lib/soundfx/libqcomvisualizer.so"
#define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "/vendor/lib/soundfx/libqcompostprocbundle.so"
#define ADM_LIBRARY_PATH "/vendor/lib/libadm.so"
#endif
/* Flags used to initialize acdb_settings variable that goes to ACDB library */
#define NONE_FLAG 0x00000000
#define ANC_FLAG 0x00000001
#define DMIC_FLAG 0x00000002
#define QMIC_FLAG 0x00000004
/* Include TMIC Flag after existing QMIC flag to avoid backward compatibility
* issues since they are bit masked */
#define TMIC_FLAG 0x00000008
#define TTY_MODE_OFF 0x00000010
#define TTY_MODE_FULL 0x00000020
#define TTY_MODE_VCO 0x00000040
#define TTY_MODE_HCO 0x00000080
#define TTY_MODE_CLEAR 0xFFFFFF0F
#define FLUENCE_MODE_CLEAR 0xFFFFFFF0
#define ACDB_DEV_TYPE_OUT 1
#define ACDB_DEV_TYPE_IN 2
/* SCO SWB codec mode */
#define SPEECH_MODE_INVALID 0xFFFF
/* support positional and index masks to 8ch */
#define MAX_SUPPORTED_CHANNEL_MASKS (2 * FCC_8)
#define MAX_SUPPORTED_FORMATS 15
#define MAX_SUPPORTED_SAMPLE_RATES 7
#define DEFAULT_HDMI_OUT_CHANNELS 2
#define DEFAULT_HDMI_OUT_SAMPLE_RATE 48000
#define DEFAULT_HDMI_OUT_FORMAT AUDIO_FORMAT_PCM_16_BIT
#define ERROR_LOG_ENTRIES 16
#define SND_CARD_STATE_OFFLINE 0
#define SND_CARD_STATE_ONLINE 1
#define STREAM_DIRECTION_IN 0
#define STREAM_DIRECTION_OUT 1
#define MAX_PERF_LOCK_OPTS 20
#define MAX_STREAM_PROFILE_STR_LEN 32
typedef enum {
EFFECT_NONE = 0,
EFFECT_AEC,
EFFECT_NS,
EFFECT_MAX
} effect_type_t;
struct audio_effect_config {
uint32_t module_id;
uint32_t instance_id;
uint32_t param_id;
uint32_t param_value;
};
struct audio_fluence_mmsecns_config {
uint32_t topology_id;
uint32_t module_id;
uint32_t instance_id;
uint32_t param_id;
};
#define MAX_MIXER_PATH_LEN 64
typedef enum card_status_t {
CARD_STATUS_OFFLINE,
CARD_STATUS_ONLINE
} card_status_t;
/* These are the supported use cases by the hardware.
* Each usecase is mapped to a specific PCM device.
* Refer to pcm_device_table[].
*/
enum {
USECASE_INVALID = -1,
/* Playback usecases */
USECASE_AUDIO_PLAYBACK_DEEP_BUFFER = 0,
USECASE_AUDIO_PLAYBACK_LOW_LATENCY,
USECASE_AUDIO_PLAYBACK_MULTI_CH,
USECASE_AUDIO_PLAYBACK_OFFLOAD,
USECASE_AUDIO_PLAYBACK_OFFLOAD2,
USECASE_AUDIO_PLAYBACK_OFFLOAD3,
USECASE_AUDIO_PLAYBACK_OFFLOAD4,
USECASE_AUDIO_PLAYBACK_OFFLOAD5,
USECASE_AUDIO_PLAYBACK_OFFLOAD6,
USECASE_AUDIO_PLAYBACK_OFFLOAD7,
USECASE_AUDIO_PLAYBACK_OFFLOAD8,
USECASE_AUDIO_PLAYBACK_OFFLOAD9,
USECASE_AUDIO_PLAYBACK_ULL,
USECASE_AUDIO_PLAYBACK_MMAP,
USECASE_AUDIO_PLAYBACK_WITH_HAPTICS,
USECASE_AUDIO_PLAYBACK_HAPTICS,
USECASE_AUDIO_PLAYBACK_HIFI,
USECASE_AUDIO_PLAYBACK_TTS,
/* FM usecase */
USECASE_AUDIO_PLAYBACK_FM,
/* HFP Use case*/
USECASE_AUDIO_HFP_SCO,
USECASE_AUDIO_HFP_SCO_WB,
USECASE_AUDIO_HFP_SCO_DOWNLINK,
USECASE_AUDIO_HFP_SCO_WB_DOWNLINK,
/* Capture usecases */
USECASE_AUDIO_RECORD,
USECASE_AUDIO_RECORD_COMPRESS,
USECASE_AUDIO_RECORD_COMPRESS2,
USECASE_AUDIO_RECORD_COMPRESS3,
USECASE_AUDIO_RECORD_COMPRESS4,
USECASE_AUDIO_RECORD_COMPRESS5,
USECASE_AUDIO_RECORD_COMPRESS6,
USECASE_AUDIO_RECORD_LOW_LATENCY,
USECASE_AUDIO_RECORD_FM_VIRTUAL,
USECASE_AUDIO_RECORD_HIFI,
USECASE_AUDIO_PLAYBACK_VOIP,
USECASE_AUDIO_RECORD_VOIP,
/* Voice usecase */
USECASE_VOICE_CALL,
USECASE_AUDIO_RECORD_MMAP,
/* Voice extension usecases */
USECASE_VOICE2_CALL,
USECASE_VOLTE_CALL,
USECASE_QCHAT_CALL,
USECASE_VOWLAN_CALL,
USECASE_VOICEMMODE1_CALL,
USECASE_VOICEMMODE2_CALL,
USECASE_COMPRESS_VOIP_CALL,
USECASE_INCALL_REC_UPLINK,
USECASE_INCALL_REC_DOWNLINK,
USECASE_INCALL_REC_UPLINK_AND_DOWNLINK,
USECASE_INCALL_REC_UPLINK_COMPRESS,
USECASE_INCALL_REC_DOWNLINK_COMPRESS,
USECASE_INCALL_REC_UPLINK_AND_DOWNLINK_COMPRESS,
USECASE_INCALL_MUSIC_UPLINK,
USECASE_INCALL_MUSIC_UPLINK2,
USECASE_AUDIO_SPKR_CALIB_RX,
USECASE_AUDIO_SPKR_CALIB_TX,
USECASE_AUDIO_PLAYBACK_AFE_PROXY,
USECASE_AUDIO_RECORD_AFE_PROXY,
USECASE_AUDIO_RECORD_AFE_PROXY2,
USECASE_AUDIO_DSM_FEEDBACK,
USECASE_AUDIO_PLAYBACK_SILENCE,
USECASE_AUDIO_TRANSCODE_LOOPBACK_RX,
USECASE_AUDIO_TRANSCODE_LOOPBACK_TX,
USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM1,
USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM2,
USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM3,
USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM4,
USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM5,
USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM6,
USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM7,
USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM8,
USECASE_AUDIO_EC_REF_LOOPBACK,
USECASE_AUDIO_A2DP_ABR_FEEDBACK,
/* car streams usecases */
USECASE_AUDIO_PLAYBACK_MEDIA,
USECASE_AUDIO_PLAYBACK_SYS_NOTIFICATION,
USECASE_AUDIO_PLAYBACK_NAV_GUIDANCE,
USECASE_AUDIO_PLAYBACK_PHONE,
USECASE_AUDIO_PLAYBACK_FRONT_PASSENGER,
USECASE_AUDIO_PLAYBACK_REAR_SEAT,
/*Audio FM Tuner usecase*/
USECASE_AUDIO_FM_TUNER_EXT,
AUDIO_USECASE_MAX
};
const char * const use_case_table[AUDIO_USECASE_MAX];
#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
/*
* tinyAlsa library interprets period size as number of frames
* one frame = channel_count * sizeof (pcm sample)
* so if format = 16-bit PCM and channels = Stereo, frame size = 2 ch * 2 = 4 bytes
* DEEP_BUFFER_OUTPUT_PERIOD_SIZE = 1024 means 1024 * 4 = 4096 bytes
* We should take care of returning proper size when AudioFlinger queries for
* the buffer size of an input/output stream
*/
enum {
OFFLOAD_CMD_EXIT, /* exit compress offload thread loop*/
OFFLOAD_CMD_DRAIN, /* send a full drain request to DSP */
OFFLOAD_CMD_PARTIAL_DRAIN, /* send a partial drain request to DSP */
OFFLOAD_CMD_WAIT_FOR_BUFFER, /* wait for buffer released by DSP */
OFFLOAD_CMD_ERROR, /* offload playback hit some error */
};
/*
* Camera selection indicated via set_parameters "cameraFacing=front|back and
* "rotation=0|90|180|270""
*/
enum {
CAMERA_FACING_BACK = 0x0,
CAMERA_FACING_FRONT = 0x1,
CAMERA_FACING_MASK = 0x0F,
CAMERA_ROTATION_LANDSCAPE = 0x0,
CAMERA_ROTATION_INVERT_LANDSCAPE = 0x10,
CAMERA_ROTATION_PORTRAIT = 0x20,
CAMERA_ROTATION_MASK = 0xF0,
CAMERA_BACK_LANDSCAPE = (CAMERA_FACING_BACK|CAMERA_ROTATION_LANDSCAPE),
CAMERA_BACK_INVERT_LANDSCAPE = (CAMERA_FACING_BACK|CAMERA_ROTATION_INVERT_LANDSCAPE),
CAMERA_BACK_PORTRAIT = (CAMERA_FACING_BACK|CAMERA_ROTATION_PORTRAIT),
CAMERA_FRONT_LANDSCAPE = (CAMERA_FACING_FRONT|CAMERA_ROTATION_LANDSCAPE),
CAMERA_FRONT_INVERT_LANDSCAPE = (CAMERA_FACING_FRONT|CAMERA_ROTATION_INVERT_LANDSCAPE),
CAMERA_FRONT_PORTRAIT = (CAMERA_FACING_FRONT|CAMERA_ROTATION_PORTRAIT),
CAMERA_DEFAULT = CAMERA_BACK_LANDSCAPE,
};
//FIXME: to be replaced by proper video capture properties API
#define AUDIO_PARAMETER_KEY_CAMERA_FACING "cameraFacing"
#define AUDIO_PARAMETER_VALUE_FRONT "front"
#define AUDIO_PARAMETER_VALUE_BACK "back"
enum {
OFFLOAD_STATE_IDLE,
OFFLOAD_STATE_PLAYING,
OFFLOAD_STATE_PAUSED,
};
struct offload_cmd {
struct listnode node;
int cmd;
int data[];
};
typedef enum render_mode {
RENDER_MODE_AUDIO_NO_TIMESTAMP = 0,
RENDER_MODE_AUDIO_MASTER,
RENDER_MODE_AUDIO_STC_MASTER,
} render_mode_t;
/* This defines the physical car audio streams supported in
* audio HAL, limited by the available frontend PCM devices.
* Max number of physical streams supported is 32 and is
* represented by stream bit flag.
* Primary zone: bit 0 - 7
* Front passenger zone: bit 8 - 15
* Rear seat zone: bit 16 - 23
*/
#define MAX_CAR_AUDIO_STREAMS 32
enum {
CAR_AUDIO_STREAM_MEDIA = 0x1,
CAR_AUDIO_STREAM_SYS_NOTIFICATION = 0x2,
CAR_AUDIO_STREAM_NAV_GUIDANCE = 0x4,
CAR_AUDIO_STREAM_PHONE = 0x8,
CAR_AUDIO_STREAM_FRONT_PASSENGER = 0x100,
CAR_AUDIO_STREAM_REAR_SEAT = 0x10000,
};
struct stream_app_type_cfg {
int sample_rate;
uint32_t bit_width;
int app_type;
int gain[2];
};
struct stream_config {
unsigned int sample_rate;
audio_channel_mask_t channel_mask;
audio_format_t format;
struct listnode device_list;
unsigned int bit_width;
};
struct stream_inout {
pthread_mutex_t lock; /* see note below on mutex acquisition order */
pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by playback thread */
pthread_cond_t cond;
struct stream_config in_config;
struct stream_config out_config;
struct stream_app_type_cfg out_app_type_cfg;
char profile[MAX_STREAM_PROFILE_STR_LEN];
struct audio_device *dev;
void *adsp_hdlr_stream_handle;
void *ip_hdlr_handle;
stream_callback_t client_callback;
void *client_cookie;
};
struct stream_out {
struct audio_stream_out stream;
pthread_mutex_t lock; /* see note below on mutex acquisition order */
pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by playback thread */
pthread_cond_t cond;
/* stream_out->lock is of large granularity, and can only be held before device lock
* latch is a supplemetary lock to protect certain fields of out stream and
* it can be held after device lock
*/
pthread_mutex_t latch_lock;
pthread_mutex_t position_query_lock; /* sychronize frame written */
struct pcm_config config;
struct compr_config compr_config;
struct pcm *pcm;
struct compress *compr;
int standby;
int pcm_device_id;
unsigned int sample_rate;
audio_channel_mask_t channel_mask;
audio_format_t format;
struct listnode device_list;
audio_output_flags_t flags;
char profile[MAX_STREAM_PROFILE_STR_LEN];
audio_usecase_t usecase;
/* Array of supported channel mask configurations. +1 so that the last entry is always 0 */
audio_channel_mask_t supported_channel_masks[MAX_SUPPORTED_CHANNEL_MASKS + 1];
audio_format_t supported_formats[MAX_SUPPORTED_FORMATS+1];
uint32_t supported_sample_rates[MAX_SUPPORTED_SAMPLE_RATES+1];
bool muted;
uint64_t written; /* total frames written, not cleared when entering standby */
int64_t mmap_time_offset_nanos; /* fudge factor to correct inaccuracies in DSP */
int mmap_shared_memory_fd; /* file descriptor associated with MMAP NOIRQ shared memory */
audio_io_handle_t handle;
struct stream_app_type_cfg app_type_cfg;
int non_blocking;
int playback_started;
int offload_state;
pthread_cond_t offload_cond;
pthread_t offload_thread;
struct listnode offload_cmd_list;
bool offload_thread_blocked;
struct timespec writeAt;
void *adsp_hdlr_stream_handle;
void *ip_hdlr_handle;
stream_callback_t client_callback;
void *client_cookie;
struct compr_gapless_mdata gapless_mdata;
int send_new_metadata;
bool send_next_track_params;
bool is_compr_metadata_avail;
unsigned int bit_width;
uint32_t hal_fragment_size;
audio_format_t hal_ip_format;
audio_format_t hal_op_format;
void *convert_buffer;
bool realtime;
int af_period_multiplier;
struct audio_device *dev;
card_status_t card_status;
void* qaf_stream_handle;
void* qap_stream_handle;
pthread_cond_t qaf_offload_cond;
pthread_t qaf_offload_thread;
struct listnode qaf_offload_cmd_list;
uint32_t platform_latency;
render_mode_t render_mode;
bool drift_correction_enabled;
struct audio_out_channel_map_param channel_map_param; /* input channel map */
audio_offload_info_t info;
int started;
qahwi_stream_out_t qahwi_out;
bool is_iec61937_info_available;
bool a2dp_muted;
float volume_l;
float volume_r;
bool apply_volume;
char pm_qos_mixer_path[MAX_MIXER_PATH_LEN];
int hal_output_suspend_supported;
int dynamic_pm_qos_config_supported;
bool stream_config_changed;
mix_matrix_params_t pan_scale_params;
mix_matrix_params_t downmix_params;
bool set_dual_mono;
bool prev_card_status_offline;
error_log_t *error_log;
bool pspd_coeff_sent;
int car_audio_stream;
union {
char *addr;
struct {
int controller;
int stream;
} cs;
} extconn;
size_t kernel_buffer_size; // cached value of the alsa buffer size, const after open().
// last out_get_presentation_position() cached info.
bool last_fifo_valid;
unsigned int last_fifo_frames_remaining;
int64_t last_fifo_time_ns;
simple_stats_t fifo_underruns; // TODO: keep a list of the last N fifo underrun times.
simple_stats_t start_latency_ms;
};
struct stream_in {
struct audio_stream_in stream;
pthread_mutex_t lock; /* see note below on mutex acquisition order */
pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by playback thread */
struct pcm_config config;
struct pcm *pcm;
int standby;
int source;
int pcm_device_id;
struct listnode device_list;
audio_channel_mask_t channel_mask;
audio_usecase_t usecase;
bool enable_aec;
bool enable_ns;
audio_format_t format;
bool enable_ec_port;
bool ec_opened;
struct listnode aec_list;
struct listnode ns_list;
int64_t mmap_time_offset_nanos; /* fudge factor to correct inaccuracies in DSP */
int mmap_shared_memory_fd; /* file descriptor associated with MMAP NOIRQ shared memory */
audio_io_handle_t capture_handle;
audio_input_flags_t flags;
char profile[MAX_STREAM_PROFILE_STR_LEN];
bool is_st_session;
bool is_st_session_active;
unsigned int sample_rate;
unsigned int bit_width;
bool realtime;
int af_period_multiplier;
struct stream_app_type_cfg app_type_cfg;
void *cin_extn;
qahwi_stream_in_t qahwi_in;
struct audio_device *dev;
card_status_t card_status;
int capture_started;
float zoom;
audio_microphone_direction_t direction;
volatile int32_t capture_stopped;
/* Array of supported channel mask configurations. +1 so that the last entry is always 0 */
audio_channel_mask_t supported_channel_masks[MAX_SUPPORTED_CHANNEL_MASKS + 1];
audio_format_t supported_formats[MAX_SUPPORTED_FORMATS + 1];
uint32_t supported_sample_rates[MAX_SUPPORTED_SAMPLE_RATES + 1];
int64_t frames_read; /* total frames read, not cleared when entering standby */
int64_t frames_muted; /* total frames muted, not cleared when entering standby */
error_log_t *error_log;
simple_stats_t start_latency_ms;
};
typedef enum {
PCM_PLAYBACK,
PCM_CAPTURE,
VOICE_CALL,
VOIP_CALL,
PCM_HFP_CALL,
TRANSCODE_LOOPBACK_RX,
TRANSCODE_LOOPBACK_TX,
PCM_PASSTHROUGH,
USECASE_TYPE_MAX
} usecase_type_t;
typedef enum {
PATCH_NONE = -1,
PATCH_PLAYBACK,
PATCH_CAPTURE,
PATCH_DEVICE_LOOPBACK
} patch_type_t;
struct audio_patch_info {
struct audio_patch *patch;
patch_type_t patch_type;
};
struct audio_stream_info {
struct audio_stream *stream;
audio_patch_handle_t patch_handle;
};
union stream_ptr {
struct stream_in *in;
struct stream_out *out;
struct stream_inout *inout;
};
struct audio_usecase {
struct listnode list;
audio_usecase_t id;
usecase_type_t type;
struct listnode device_list;
snd_device_t out_snd_device;
snd_device_t in_snd_device;
struct stream_app_type_cfg out_app_type_cfg;
struct stream_app_type_cfg in_app_type_cfg;
union stream_ptr stream;
};
struct stream_format {
struct listnode list;
audio_format_t format;
};
struct stream_sample_rate {
struct listnode list;
uint32_t sample_rate;
};
typedef union {
audio_output_flags_t out_flags;
audio_input_flags_t in_flags;
} audio_io_flags_t;
struct streams_io_cfg {
struct listnode list;
audio_io_flags_t flags;
char profile[MAX_STREAM_PROFILE_STR_LEN];
struct listnode format_list;
struct listnode sample_rate_list;
struct stream_app_type_cfg app_type_cfg;
};
typedef struct streams_input_ctxt {
struct listnode list;
struct stream_in *input;
} streams_input_ctxt_t;
typedef struct streams_output_ctxt {
struct listnode list;
struct stream_out *output;
} streams_output_ctxt_t;
typedef void* (*adm_init_t)();
typedef void (*adm_deinit_t)(void *);
typedef void (*adm_register_output_stream_t)(void *, audio_io_handle_t, audio_output_flags_t);
typedef void (*adm_register_input_stream_t)(void *, audio_io_handle_t, audio_input_flags_t);
typedef void (*adm_deregister_stream_t)(void *, audio_io_handle_t);
typedef void (*adm_request_focus_t)(void *, audio_io_handle_t);
typedef void (*adm_abandon_focus_t)(void *, audio_io_handle_t);
typedef void (*adm_set_config_t)(void *, audio_io_handle_t,
struct pcm *,
struct pcm_config *);
typedef void (*adm_request_focus_v2_t)(void *, audio_io_handle_t, long);
typedef bool (*adm_is_noirq_avail_t)(void *, int, int, int);
typedef void (*adm_on_routing_change_t)(void *, audio_io_handle_t);
typedef int (*adm_request_focus_v2_1_t)(void *, audio_io_handle_t, long);
struct audio_device {
struct audio_hw_device device;
pthread_mutex_t lock; /* see note below on mutex acquisition order */
pthread_mutex_t cal_lock;
struct mixer *mixer;
audio_mode_t mode;
audio_mode_t prev_mode;
audio_devices_t out_device;
struct stream_out *primary_output;
struct stream_out *voice_tx_output;
struct stream_out *current_call_output;
bool bluetooth_nrec;
bool screen_off;
int *snd_dev_ref_cnt;
struct listnode usecase_list;
struct listnode streams_output_cfg_list;
struct listnode streams_input_cfg_list;
struct audio_route *audio_route;
int acdb_settings;
bool speaker_lr_swap;
struct voice voice;
unsigned int cur_hdmi_channels;
audio_format_t cur_hdmi_format;
unsigned int cur_hdmi_sample_rate;
unsigned int cur_hdmi_bit_width;
unsigned int cur_wfd_channels;
bool bt_wb_speech_enabled;
unsigned int swb_speech_mode;
bool allow_afe_proxy_usage;
bool is_charging; // from battery listener
bool mic_break_enabled;
bool enable_hfp;
bool mic_muted;
bool enable_voicerx;
unsigned int num_va_sessions;
int snd_card;
card_status_t card_status;
unsigned int cur_codec_backend_samplerate;
unsigned int cur_codec_backend_bit_width;
bool is_channel_status_set;
void *platform;
void *extspk;
unsigned int offload_usecases_state;
unsigned int pcm_record_uc_state;
void *visualizer_lib;
int (*visualizer_start_output)(audio_io_handle_t, int);
int (*visualizer_stop_output)(audio_io_handle_t, int);
void *offload_effects_lib;
int (*offload_effects_start_output)(audio_io_handle_t, int, struct mixer *);
int (*offload_effects_stop_output)(audio_io_handle_t, int);
int (*offload_effects_set_hpx_state)(bool);
void *adm_data;
void *adm_lib;
adm_init_t adm_init;
adm_deinit_t adm_deinit;
adm_register_input_stream_t adm_register_input_stream;
adm_register_output_stream_t adm_register_output_stream;
adm_deregister_stream_t adm_deregister_stream;
adm_request_focus_t adm_request_focus;
adm_abandon_focus_t adm_abandon_focus;
adm_set_config_t adm_set_config;
adm_request_focus_v2_t adm_request_focus_v2;
adm_is_noirq_avail_t adm_is_noirq_avail;
adm_on_routing_change_t adm_on_routing_change;
adm_request_focus_v2_1_t adm_request_focus_v2_1;
void (*offload_effects_get_parameters)(struct str_parms *,
struct str_parms *);
void (*offload_effects_set_parameters)(struct str_parms *);
bool multi_offload_enable;
int perf_lock_handle;
int perf_lock_opts[MAX_PERF_LOCK_OPTS];
int perf_lock_opts_size;
bool native_playback_enabled;
bool asrc_mode_enabled;
qahwi_device_t qahwi_dev;
bool vr_audio_mode_enabled;
uint32_t dsp_bit_width_enforce_mode;
bool bt_sco_on;
struct audio_device_config_param *device_cfg_params;
unsigned int interactive_usecase_state;
bool dp_allowed_for_voice;
void *ext_hw_plugin;
struct pcm_config haptics_config;
struct pcm *haptic_pcm;
int haptic_pcm_device_id;
uint8_t *haptic_buffer;
size_t haptic_buffer_size;
/* logging */
snd_device_t last_logged_snd_device[AUDIO_USECASE_MAX][2]; /* [out, in] */
/* The pcm_params use_case_table is loaded by adev_verify_devices() upon
* calling adev_open().
*
* If an entry is not NULL, it can be used to determine if extended precision
* or other capabilities are present for the device corresponding to that usecase.
*/
struct pcm_params *use_case_table[AUDIO_USECASE_MAX];
struct listnode active_inputs_list;
struct listnode active_outputs_list;
bool use_old_pspd_mix_ctrl;
int camera_orientation; /* CAMERA_BACK_LANDSCAPE ... CAMERA_FRONT_PORTRAIT */
bool adm_routing_changed;
struct listnode audio_patch_record_list;
Hashmap *patch_map;
Hashmap *io_streams_map;
bool a2dp_started;
bool ha_proxy_enable;
};
struct audio_patch_record {
struct listnode list;
audio_patch_handle_t handle;
audio_usecase_t usecase;
struct audio_patch patch;
};
int select_devices(struct audio_device *adev,
audio_usecase_t uc_id);
int disable_audio_route(struct audio_device *adev,
struct audio_usecase *usecase);
int disable_snd_device(struct audio_device *adev,
snd_device_t snd_device);
int enable_snd_device(struct audio_device *adev,
snd_device_t snd_device);
int enable_audio_route(struct audio_device *adev,
struct audio_usecase *usecase);
struct audio_usecase *get_usecase_from_list(const struct audio_device *adev,
audio_usecase_t uc_id);
bool is_offload_usecase(audio_usecase_t uc_id);
bool audio_is_true_native_stream_active(struct audio_device *adev);
bool audio_is_dsd_native_stream_active(struct audio_device *adev);
uint32_t adev_get_dsp_bit_width_enforce_mode();
int pcm_ioctl(struct pcm *pcm, int request, ...);
audio_usecase_t get_usecase_id_from_usecase_type(const struct audio_device *adev,
usecase_type_t type);
int check_a2dp_restore_l(struct audio_device *adev, struct stream_out *out, bool restore);
int adev_open_output_stream(struct audio_hw_device *dev,
audio_io_handle_t handle,
audio_devices_t devices,
audio_output_flags_t flags,
struct audio_config *config,
struct audio_stream_out **stream_out,
const char *address);
void adev_close_output_stream(struct audio_hw_device *dev __unused,
struct audio_stream_out *stream);
bool is_interactive_usecase(audio_usecase_t uc_id);
streams_input_ctxt_t *in_get_stream(struct audio_device *dev,
audio_io_handle_t input);
streams_output_ctxt_t *out_get_stream(struct audio_device *dev,
audio_io_handle_t output);
size_t get_output_period_size(uint32_t sample_rate,
audio_format_t format,
int channel_count,
int duration /*in millisecs*/);
#define LITERAL_TO_STRING(x) #x
#define CHECK(condition) LOG_ALWAYS_FATAL_IF(!(condition), "%s",\
__FILE__ ":" LITERAL_TO_STRING(__LINE__)\
" ASSERT_FATAL(" #condition ") failed.")
static inline bool is_loopback_input_device(audio_devices_t device) {
if (!audio_is_output_device(device) &&
((device & AUDIO_DEVICE_IN_LOOPBACK) == AUDIO_DEVICE_IN_LOOPBACK))
return true;
else
return false;
}
static inline bool audio_is_virtual_input_source(audio_source_t source) {
bool result = false;
switch(source) {
case AUDIO_SOURCE_VOICE_UPLINK :
case AUDIO_SOURCE_VOICE_DOWNLINK :
case AUDIO_SOURCE_VOICE_CALL :
case AUDIO_SOURCE_FM_TUNER :
result = true;
break;
default:
break;
}
return result;
}
int route_output_stream(struct stream_out *stream,
struct listnode *devices);
int route_input_stream(struct stream_in *stream,
struct listnode *devices,
audio_source_t source);
audio_patch_handle_t generate_patch_handle();
/*
* NOTE: when multiple mutexes have to be acquired, always take the
* stream_in or stream_out mutex first, followed by the audio_device mutex.
*/
#endif // QCOM_AUDIO_HW_H