add some code
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#pragma once
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#include "esp_aec.h"
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#include "esp_agc.h"
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#include "esp_nsn_models.h"
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#include "esp_vad.h"
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#include "esp_vadn_models.h"
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#include "esp_wn_iface.h"
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#include "esp_wn_models.h"
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#include "model_path.h"
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#include "stdbool.h"
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#include "stdint.h"
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#include "stdlib.h"
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#ifdef __cplusplus
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extern "C" {
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#endif
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// AFE: Audio Front-End
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// SR: Speech Recognition
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// VC: Voice Communication
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// Set AFE_SR mode
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typedef enum {
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SR_MODE_LOW_COST = 0, // Deprecated, please use afe_mode_t, AFE mode: low cost mode
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SR_MODE_HIGH_PERF = 1, // Deprecated, please use afe_mode_t, AFE mode: high performance mode
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} afe_sr_mode_t;
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// Set AFE mode
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typedef enum {
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AFE_MODE_LOW_COST = 0, // AFE mode: low cost mode
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AFE_MODE_HIGH_PERF = 1, // AFE mode: high performance mode
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} afe_mode_t;
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// Set AFE type
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typedef enum {
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AFE_TYPE_SR = 0, // Speech recognition scenarios, excluding nonlinear noise suppression
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AFE_TYPE_VC = 1, // Voice communication scenarios, 16KHz input, including nonlinear noise suppression
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AFE_TYPE_VC_8K = 2, // Voice communication scenarios, 8KHz input, note that the input data must be 8KHz
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} afe_type_t;
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typedef enum {
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AFE_MEMORY_ALLOC_MORE_INTERNAL = 1, // malloc with more internal ram
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AFE_MEMORY_ALLOC_INTERNAL_PSRAM_BALANCE = 2, // malloc with internal ram and psram in balance
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AFE_MEMORY_ALLOC_MORE_PSRAM = 3 // malloc with more psram
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} afe_memory_alloc_mode_t;
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typedef enum {
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AFE_MN_PEAK_AGC_MODE_1 = -9, // The peak amplitude of fetch audio is -9dB
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AFE_MN_PEAK_AGC_MODE_2 = -6, // The peak amplitude of fetch audio is -6dB
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AFE_MN_PEAK_AGC_MODE_3 = -3, // The peak amplitude of fetcg is -3dB
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AFE_MN_PEAK_NO_AGC = 0, // There is no agc gain
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} afe_mn_peak_agc_mode_t;
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typedef struct {
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int total_ch_num; // total channel num, include microphone channel, playback channel and unknown channel
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int mic_num; // microphone channel number
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uint8_t *mic_ids; // microphone channel indices
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int ref_num; // playback reference channel number
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uint8_t *ref_ids; // playback reference channel indices
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int sample_rate; // sample rate of audio
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} afe_pcm_config_t;
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typedef enum {
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AFE_NS_MODE_WEBRTC = 0, // please use model name of NS, SSP: "WEBRTC"
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AFE_NS_MODE_NET = 1, // please use model name of NSNET
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} afe_ns_mode_t;
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typedef enum {
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AFE_AGC_MODE_WEBRTC = 0, // WEBRTC AGC
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AFE_AGC_MODE_WAKENET = 1, // AGC gain is calculated by wakenet model if wakenet is activated
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} afe_agc_mode_t;
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/**
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* @brief Function to get the debug audio data
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*
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* @param data The debug audio data which don't be modify. It should be copied away as soon as possible that
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* avoid blocking for too long.
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* @param data_size The number of bytes of data.
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* @returns
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*/
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typedef void (*afe_debug_hook_callback_t)(const int16_t *data, int data_size);
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typedef enum {
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AFE_DEBUG_HOOK_MASE_TASK_IN = 0, // To get the input data of mase task
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AFE_DEBUG_HOOK_FETCH_TASK_IN = 1, // To get the input data of fetch task
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AFE_DEBUG_HOOK_MAX = 2
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} afe_debug_hook_type_t;
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typedef struct {
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afe_debug_hook_type_t hook_type; // debug type of hook
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afe_debug_hook_callback_t hook_callback; // callback function which transfer debug audio data
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} afe_debug_hook_t;
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typedef struct {
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/********** AEC(Acoustic Echo Cancellation) **********/
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bool aec_init; // Whether to init aec
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aec_mode_t aec_mode; // The mode of aec, AEC_MODE_SR_LOW_COST or AEC_MODE_SR_HIGH_PERF
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int aec_filter_length; // The filter length of aec
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/********** SE(Speech Enhancement, microphone array processing) **********/
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bool se_init; // Whether to init se
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/********** NS(Noise Suppression) **********/
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bool ns_init; // Whether to init ns
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char *ns_model_name; // Model name of ns
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afe_ns_mode_t afe_ns_mode; // Model mode of ns
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/********** VAD(Voice Activity Detection) **********/
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bool vad_init; // Whether to init vad
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vad_mode_t vad_mode; // The value can be: VAD_MODE_0, VAD_MODE_1, VAD_MODE_2, VAD_MODE_3, VAD_MODE_4
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char *vad_model_name; // The model name of vad, If it is null, WebRTC VAD will be used.
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int vad_min_speech_ms; // The minimum duration of speech in ms. It should be bigger than 32 ms, default: 128 ms
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int vad_min_noise_ms; // The minimum duration of noise or silence in ms. It should be bigger than 64 ms, default:
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// 1000 ms
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int vad_delay_ms; // The delay of the first speech frame in ms, default: 128 ms
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// If you find vad cache can not cover all speech, please increase this value.
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bool vad_mute_playback; // If true, the playback will be muted for vad detection. default: false
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bool vad_enable_channel_trigger; // If true, the vad will be used to choose the channel id. default: false
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/********** WakeNet(Wake Word Engine) **********/
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bool wakenet_init;
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char *wakenet_model_name; // The model name of wakenet 1
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char *wakenet_model_name_2; // The model name of wakenet 2 if has wakenet 2
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det_mode_t wakenet_mode; // The mode of wakenet
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/********** AGC(Automatic Gain Control) **********/
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bool agc_init; // Whether to init agc
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afe_agc_mode_t
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agc_mode; // The AGC mode for ASR. and the gain generated by AGC acts on the audio after far linear gain.
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int agc_compression_gain_db; // Compression gain in dB (default 9)
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int agc_target_level_dbfs; // Target level in -dBfs of envelope (default 3, means target level is -3 dBFS)
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/********** General AFE(Audio Front End) parameter **********/
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afe_pcm_config_t pcm_config; // Config the channel num of original data which is fed to the afe feed function.
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afe_mode_t afe_mode; // The mode of afe， AFE_MODE_LOW_COST or AFE_MODE_HIGH_PERF
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afe_type_t afe_type; // The mode of afe， AFE_MODE_LOW_COST or AFE_MODE_HIGH_PERF
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int afe_perferred_core; // The preferred core of afe se task, which is created in afe_create function.
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int afe_perferred_priority; // The preferred priority of afe se task, which is created in afe_create function.
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int afe_ringbuf_size; // The ring buffer size: the number of frame data in ring buffer.
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afe_memory_alloc_mode_t memory_alloc_mode; // The memory alloc mode for afe. From Internal RAM or PSRAM
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float afe_linear_gain; // The linear gain for afe output the value should be in [0.1, 10.0]. This value acts
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// directly on the output amplitude: out_linear_gain * amplitude.
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bool debug_init;
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bool fixed_first_channel; // If true, the channel after first wake-up is fixed to raw data of microphone
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// otherwise, select channel number by wakenet
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} afe_config_t;
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/**
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* @brief Get AFE default configuration. The default configuration will enable all algorithms as much as possible based
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* on the chip target and input format. You can manually fine-tune it after creating the configuration
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*
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* The input format:
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* M to represent the microphone channel
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* R to represent the playback reference channel
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* N to represent an unknown or unused channel
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*
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* For example, input_format="MMNR" indicates that the input data consists of four channels,
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* which are the microphone channel, the microphone channel, an unused channel, and the playback channel
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*
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* @param input_format The input format
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* @param models Models from partition, which is configured by Kconfig
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* @param type The type of afe, AFE_TYPE_SR or AFE_TYPE_VC
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* @param mode The mode of afe, AFE_MODE_LOW_COST or AFE_MODE_HIGH_PERF
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*
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* @return afe_config_t* The default config of afe
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*/
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afe_config_t *afe_config_init(const char *input_format, srmodel_list_t *models, afe_type_t type, afe_mode_t mode);
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/**
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* @brief Check AFE configuration and make sure it is correct.
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*
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* @warning If there is a configuration conflict, this function will modify some parameters.
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* The guiding behind these modifications is to maintain the highest performance of the output audio and results.
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* And remove the conflict between different algorithms.
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*
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* For example, If input is two-channel data, the SE(BSS) algorithm will be prioritized over the NS algorithm.
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* If SE(BSS) algorithm is deactivated, will only use the first microphone channel.
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*
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* @param afe_config Input AFE config
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*
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* @return afe_config_t* The modified AFE config
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*/
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afe_config_t *afe_config_check(afe_config_t *afe_config);
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/**
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* @brief Parse input format
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*
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* @param input_format The input format, same with afe_config_init() function
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* @param pcm_config The pcm config
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*
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* @return true if the input format is parsed successfully, otherwise false
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*/
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bool afe_parse_input_format(const char *input_format, afe_pcm_config_t *pcm_config);
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/**
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* @brief Parse I2S input data
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*
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* @param data The input multi channel data
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* @param frame_size The frame size of input, it is also the size of single channel data
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* @param mic_data The output microphone data
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* @param ref_data The output playback reference data
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* @param pcm_config The pcm config
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*
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*/
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void afe_parse_input(int16_t *data, int frame_size, int16_t *mic_data, int16_t *ref_data, afe_pcm_config_t *pcm_config);
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/**
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* @brief Parse input data, from interleaved arrangement to contiguous arrangement
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*
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* @param data The input multi channel data
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* @param frame_size The frame size of input, it is also the size of single channel data
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* @param channel_num The channel number of data
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* @param out_data The output data
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*
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*/
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void afe_parse_data(int16_t *data, int frame_size, int channel_num, int16_t *out_data);
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/**
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* @brief Format input data, from contiguous arrangement to interleaved arrangement
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*
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* @param data The input multi channel data
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* @param frame_size The frame size of input, it is also the size of single channel data
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* @param channel_num The channel number of data
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* @param out_data The output data
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*
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*/
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void afe_format_data(int16_t *data, int frame_size, int channel_num, int16_t *out_data);
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/**
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* @brief Adjust the gain of input data
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*
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* @warning the input data will be modified inplace.
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*
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* @param data The input audio data
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* @param frame_size The frame size of input, it is also the size of single channel data
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* @param factor The gain factor
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*
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* @return int16_t* The output audio data
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*/
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int16_t *afe_adjust_gain(int16_t *data, int frame_size, float factor);
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/**
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* @brief Adjust the gain of input data
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*
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* @warning the input data will be modified inplace.
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*
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* @param in_data The input audio data
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* @param in_frame_size Input data frame size of input
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* @param channel_num The channel number of input data, which is same as output data
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* @param out_data The output audio data
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* @param out_frame_size Onput data frame size of input
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*
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*/
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void afe_concat_data(int16_t *in_data, int in_frame_size, int channel_num, int16_t *out_data, int out_frame_size);
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/**
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* @brief Copy the afe config
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*
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* @param dst_config The destination afe config
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* @param src_config The source afe config
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*
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* @return The destination afe config
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*/
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afe_config_t *afe_config_copy(afe_config_t *dst_config, const afe_config_t *src_config);
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/**
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* @brief Print the afe config
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*
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* @param afe_config The afe config
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*/
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void afe_config_print(const afe_config_t *afe_config);
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/**
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* @brief Allocate afe config
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*
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* @return The afe config pointer
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*/
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afe_config_t *afe_config_alloc();
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/**
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* @brief Free afe config
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*
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* @param afe_config The afe config pointer
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*/
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void afe_config_free(afe_config_t *afe_config);
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#ifdef __cplusplus
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}
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#endif
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