hal/audio_hw.h - LeafOS-Project/android_hardware_qcom_audio - Gitiles

 /*
  * Copyright (c) 2013-2018, 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/list.h>
 #include <hardware/audio.h>
 #include <tinyalsa/asoundlib.h>
 #include <tinycompress/tinycompress.h>

 #include <audio_route/audio_route.h>
 #include "audio_defs.h"
 #include "voice.h"
 #include "audio_hw_extn_api.h"

 #if LINUX_ENABLED
 #define VISUALIZER_LIBRARY_PATH "libqcomvisualizer.so"
 #define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "libqcompostprocbundle.so"
 #define ADM_LIBRARY_PATH "libadm.so"
 #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

 #define MAX_SUPPORTED_CHANNEL_MASKS 14
 #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 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;
 };

 #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_HIFI,

     /* FM usecase */
     USECASE_AUDIO_PLAYBACK_FM,

     /* HFP Use case*/
     USECASE_AUDIO_HFP_SCO,
     USECASE_AUDIO_HFP_SCO_WB,

     /* 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_PLAYBACK_SILENCE,

     USECASE_AUDIO_TRANSCODE_LOOPBACK,

     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,

     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[];
 };

 typedef enum render_mode {
     RENDER_MODE_AUDIO_NO_TIMESTAMP = 0,
     RENDER_MODE_AUDIO_MASTER,
     RENDER_MODE_AUDIO_STC_MASTER,
 } render_mode_t;

 struct stream_app_type_cfg {
     int sample_rate;
     uint32_t bit_width;
     int app_type;
 };

 struct stream_config {
     unsigned int sample_rate;
     audio_channel_mask_t channel_mask;
     audio_format_t format;
     audio_devices_t devices;
     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_mutex_t compr_mute_lock; /* acquire before setting compress volume */
     pthread_mutex_t position_query_lock; /* acquire before updating/getting position of track offload*/
     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;
     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 */
     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;
     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_compress_mute;
     float volume_l;
     float volume_r;

     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;
 };

 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;
     audio_devices_t device;
     audio_channel_mask_t channel_mask;
     audio_usecase_t usecase;
     bool enable_aec;
     bool enable_ns;
     audio_format_t format;
     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;

     /* 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];
 };

 typedef enum {
     PCM_PLAYBACK,
     PCM_CAPTURE,
     VOICE_CALL,
     VOIP_CALL,
     PCM_HFP_CALL,
     TRANSCODE_LOOPBACK
 } usecase_type_t;

 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;
     audio_devices_t devices;
     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 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;
     audio_devices_t out_device;
     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 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;
     bool allow_afe_proxy_usage;

     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;
     unsigned int offload_usecases_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;

     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;
 };

 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);

 struct stream_in *get_next_active_input(const struct audio_device *adev);

 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(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 __unused);
 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);

 #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
	/*
	* Copyright (c) 2013-2018, 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/list.h>
	#include <hardware/audio.h>
	#include <tinyalsa/asoundlib.h>
	#include <tinycompress/tinycompress.h>

	#include <audio_route/audio_route.h>
	#include "audio_defs.h"
	#include "voice.h"
	#include "audio_hw_extn_api.h"

	#if LINUX_ENABLED
	#define VISUALIZER_LIBRARY_PATH "libqcomvisualizer.so"
	#define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "libqcompostprocbundle.so"
	#define ADM_LIBRARY_PATH "libadm.so"
	#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

	#define MAX_SUPPORTED_CHANNEL_MASKS 14
	#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 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;
	};

	#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_HIFI,

	/* FM usecase */
	USECASE_AUDIO_PLAYBACK_FM,

	/* HFP Use case*/
	USECASE_AUDIO_HFP_SCO,
	USECASE_AUDIO_HFP_SCO_WB,

	/* 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_PLAYBACK_SILENCE,

	USECASE_AUDIO_TRANSCODE_LOOPBACK,

	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,

	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[];
	};

	typedef enum render_mode {
	RENDER_MODE_AUDIO_NO_TIMESTAMP = 0,
	RENDER_MODE_AUDIO_MASTER,
	RENDER_MODE_AUDIO_STC_MASTER,
	} render_mode_t;

	struct stream_app_type_cfg {
	int sample_rate;
	uint32_t bit_width;
	int app_type;
	};

	struct stream_config {
	unsigned int sample_rate;
	audio_channel_mask_t channel_mask;
	audio_format_t format;
	audio_devices_t devices;
	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_mutex_t compr_mute_lock; /* acquire before setting compress volume */
	pthread_mutex_t position_query_lock; /* acquire before updating/getting position of track offload*/
	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;
	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 */
	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;
	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_compress_mute;
	float volume_l;
	float volume_r;

	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;
	};

	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;
	audio_devices_t device;
	audio_channel_mask_t channel_mask;
	audio_usecase_t usecase;
	bool enable_aec;
	bool enable_ns;
	audio_format_t format;
	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;

	/* 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];
	};

	typedef enum {
	PCM_PLAYBACK,
	PCM_CAPTURE,
	VOICE_CALL,
	VOIP_CALL,
	PCM_HFP_CALL,
	TRANSCODE_LOOPBACK
	} usecase_type_t;

	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;
	audio_devices_t devices;
	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 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;
	audio_devices_t out_device;
	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 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;
	bool allow_afe_proxy_usage;

	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;
	unsigned int offload_usecases_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;

	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;
	};

	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);

	struct stream_in get_next_active_input(const struct audio_device adev);

	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(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 __unused);
	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);

	#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