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PL286/hardware/qcom/audio/hal/audio_hw.h

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/*
* Copyright (C) 2013-2016 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
#ifndef QCOM_AUDIO_HW_H
#define QCOM_AUDIO_HW_H
#include <cutils/str_parms.h>
#include <cutils/list.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 "voice.h"
// dlopen() does not go through default library path search if there is a "/" in the library name.
#ifdef __LP64__
#define VISUALIZER_LIBRARY_PATH "/vendor/lib64/soundfx/libqcomvisualizer.so"
#define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "/vendor/lib64/soundfx/libqcompostprocbundle.so"
#else
#define VISUALIZER_LIBRARY_PATH "/vendor/lib/soundfx/libqcomvisualizer.so"
#define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "/vendor/lib/soundfx/libqcompostprocbundle.so"
#endif
#define ADM_LIBRARY_PATH "libadm.so"
/* Flags used to initialize acdb_settings variable that goes to ACDB library */
#define DMIC_FLAG 0x00000002
#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 ACDB_DEV_TYPE_OUT 1
#define ACDB_DEV_TYPE_IN 2
#define MAX_SUPPORTED_CHANNEL_MASKS 8
#define MAX_SUPPORTED_FORMATS 15
#define MAX_SUPPORTED_SAMPLE_RATES 7
#define DEFAULT_HDMI_OUT_CHANNELS 2
#define ERROR_LOG_ENTRIES 16
/* Error types for the error log */
enum {
ERROR_CODE_STANDBY = 1,
ERROR_CODE_WRITE,
ERROR_CODE_READ,
};
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_HIFI,
USECASE_AUDIO_PLAYBACK_OFFLOAD,
USECASE_AUDIO_PLAYBACK_TTS,
USECASE_AUDIO_PLAYBACK_ULL,
USECASE_AUDIO_PLAYBACK_MMAP,
/* HFP Use case*/
USECASE_AUDIO_HFP_SCO,
USECASE_AUDIO_HFP_SCO_WB,
/* Capture usecases */
USECASE_AUDIO_RECORD,
USECASE_AUDIO_RECORD_LOW_LATENCY,
USECASE_AUDIO_RECORD_MMAP,
USECASE_AUDIO_RECORD_HIFI,
/* Voice extension usecases
*
* Following usecase are specific to voice session names created by
* MODEM and APPS on 8992/8994/8084/8974 platforms.
*/
USECASE_VOICE_CALL, /* Usecase setup for voice session on first subscription for DSDS/DSDA */
USECASE_VOICE2_CALL, /* Usecase setup for voice session on second subscription for DSDS/DSDA */
USECASE_VOLTE_CALL, /* Usecase setup for VoLTE session on first subscription */
USECASE_QCHAT_CALL, /* Usecase setup for QCHAT session */
USECASE_VOWLAN_CALL, /* Usecase setup for VoWLAN session */
/*
* Following usecase are specific to voice session names created by
* MODEM and APPS on 8996 platforms.
*/
USECASE_VOICEMMODE1_CALL, /* Usecase setup for Voice/VoLTE/VoWLAN sessions on first
* subscription for DSDS/DSDA
*/
USECASE_VOICEMMODE2_CALL, /* Usecase setup for voice/VoLTE/VoWLAN sessions on second
* subscription for DSDS/DSDA
*/
USECASE_INCALL_REC_UPLINK,
USECASE_INCALL_REC_DOWNLINK,
USECASE_INCALL_REC_UPLINK_AND_DOWNLINK,
USECASE_AUDIO_SPKR_CALIB_RX,
USECASE_AUDIO_SPKR_CALIB_TX,
USECASE_AUDIO_PLAYBACK_AFE_PROXY,
USECASE_AUDIO_RECORD_AFE_PROXY,
USECASE_AUDIO_DSM_FEEDBACK,
/* VOIP usecase*/
USECASE_AUDIO_PLAYBACK_VOIP,
USECASE_AUDIO_RECORD_VOIP,
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 */
};
enum {
OFFLOAD_STATE_IDLE,
OFFLOAD_STATE_PLAYING,
OFFLOAD_STATE_PAUSED,
};
struct offload_cmd {
struct listnode node;
int cmd;
int data[];
};
struct stream_app_type_cfg {
int sample_rate;
uint32_t bit_width; // unused
const char *mode;
int app_type;
int gain[2];
};
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;
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;
audio_devices_t devices;
audio_output_flags_t flags;
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 */
audio_io_handle_t handle;
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;
stream_callback_t offload_callback;
void *offload_cookie;
struct compr_gapless_mdata gapless_mdata;
int send_new_metadata;
bool realtime;
int af_period_multiplier;
struct audio_device *dev;
card_status_t card_status;
error_log_t *error_log;
struct stream_app_type_cfg app_type_cfg;
};
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 capture thread */
struct pcm_config config;
struct pcm *pcm;
int standby;
int source;
int pcm_device_id;
audio_devices_t device;
audio_channel_mask_t channel_mask;
unsigned int sample_rate;
audio_usecase_t usecase;
bool enable_aec;
bool enable_ns;
int64_t frames_read; /* total frames read, not cleared when entering standby */
int64_t frames_muted; /* total frames muted, not cleared when entering standby */
audio_io_handle_t capture_handle;
audio_input_flags_t flags;
bool is_st_session;
bool is_st_session_active;
bool realtime;
int af_period_multiplier;
struct audio_device *dev;
audio_format_t format;
card_status_t card_status;
int capture_started;
struct stream_app_type_cfg app_type_cfg;
/* 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];
error_log_t *error_log;
};
typedef enum usecase_type_t {
PCM_PLAYBACK,
PCM_CAPTURE,
VOICE_CALL,
PCM_HFP_CALL,
USECASE_TYPE_MAX
} usecase_type_t;
union stream_ptr {
struct stream_in *in;
struct stream_out *out;
};
struct audio_usecase {
struct listnode list;
audio_usecase_t id;
usecase_type_t type;
audio_devices_t devices;
snd_device_t out_snd_device;
snd_device_t in_snd_device;
union stream_ptr stream;
};
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);
struct audio_device {
struct audio_hw_device device;
pthread_mutex_t lock; /* see note below on mutex acquisition order */
struct mixer *mixer;
audio_mode_t mode;
struct stream_in *active_input;
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 audio_route *audio_route;
int acdb_settings;
struct voice voice;
unsigned int cur_hdmi_channels;
bool bt_wb_speech_enabled;
bool mic_muted;
bool enable_voicerx;
bool enable_hfp;
int snd_card;
void *platform;
void *extspk;
card_status_t card_status;
void *visualizer_lib;
int (*visualizer_start_output)(audio_io_handle_t, int);
int (*visualizer_stop_output)(audio_io_handle_t, int);
/* 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];
void *offload_effects_lib;
int (*offload_effects_start_output)(audio_io_handle_t, int);
int (*offload_effects_stop_output)(audio_io_handle_t, int);
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;
/* logging */
snd_device_t last_logged_snd_device[AUDIO_USECASE_MAX][2]; /* [out, in] */
};
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(struct audio_device *adev,
audio_usecase_t uc_id);
#define LITERAL_TO_STRING(x) #x
#define CHECK(condition) LOG_ALWAYS_FATAL_IF(!(condition), "%s",\
__FILE__ ":" LITERAL_TO_STRING(__LINE__)\
" ASSERT_FATAL(" #condition ") failed.")
/*
* 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