Audio Module

Supported ICs: [RTL8735C]

Overview

The audio_module provides AP-side audio capture and playback through the built-in audio hardware. It supports both analog microphone (AMIC) and digital microphone (DMIC) inputs, and drives speaker/headphone output through the DAC.

Within the Multimedia Framework, the audio module acts as a source (capture) for upstream processing such as AAC encoding, and as a sink (playback) for decoded audio from the network or local playback.

[Audio Source] --> [audio_module] --> SISO --> [aac_module] --> ...
   (capture)       (DMA/ALC)

... --> [aad_module] --> SISO --> [audio_module] --> [Speaker/Headphone]
                                                      (playback)

audio_params_t audio_params;
mm_module_ctrl(audio_ctx, CMD_AUDIO_GET_PARAMS, (int)&audio_params);
audio_params.sample_rate    = ASR_16KHZ;
audio_params.channel        = 1;
audio_params.use_mic_type   = USE_AUDIO_AMIC;
audio_params.mic_gain       = MIC_20DB;
audio_params.avsync_en      = 1;
mm_module_ctrl(audio_ctx, CMD_AUDIO_SET_PARAMS, (int)&audio_params);
mm_module_ctrl(audio_ctx, MM_CMD_SET_QUEUE_LEN, 6);
mm_module_ctrl(audio_ctx, MM_CMD_INIT_QUEUE_ITEMS, MMQI_FLAG_STATIC);
mm_module_ctrl(audio_ctx, CMD_AUDIO_APPLY, 0);

sample_rate

Selects the audio sample rate.

Constant	Value	Description
`ASR_8KHZ`	0	8000 Hz, narrowband voice
`ASR_16KHZ`	1	16000 Hz, wideband voice
`ASR_32KHZ`	2	32000 Hz
`ASR_44p1KHZ`	3	44100 Hz, CD quality
`ASR_48KHZ`	4	48000 Hz
`ASR_88p2KHZ`	5	88200 Hz
`ASR_96KHZ`	6	96000 Hz

Note

ASP algorithms (AEC, NS, AGC) only support 8 kHz and 16 kHz. Higher sample rates bypass ASP processing.

word_length

Constant	Value	Description
`WL_16BIT`	0	16-bit PCM (default)
`WL_24BIT`	1	24-bit PCM

channel

Number of channels: 1 (mono) or 2 (stereo). When using USE_AUDIO_STEREO_DMIC, the two channels are merged into a single mono stream by default.

use_mic_type

Selects the microphone input type.

Constant	Value	Description
`USE_AUDIO_AMIC`	0	Analog microphone
`USE_AUDIO_LEFT_DMIC`	1	Left digital microphone
`USE_AUDIO_RIGHT_DMIC`	2	Right digital microphone
`USE_AUDIO_STEREO_DMIC`	3	Stereo digital microphone

mic_gain

Analog microphone input gain (only applies when use_mic_type = USE_AUDIO_AMIC).

Constant	Value	Description
`MIC_0DB`	0	0 dB gain
`MIC_20DB`	1	20 dB gain
`MIC_30DB`	2	30 dB gain
`MIC_40DB`	3	40 dB gain (default)

dmic_l_gain / dmic_r_gain

Digital microphone boost gain (only applies when using DMIC). Left and right channels are configured independently.

Constant	Value	Description
`DMIC_BOOST_0DB`	0	0 dB gain (default)
`DMIC_BOOST_12DB`	1	12 dB gain
`DMIC_BOOST_24DB`	2	24 dB gain
`DMIC_BOOST_36DB`	3	36 dB gain

ADC_gain

ADC digital volume controls the input (analog-to-digital) gain. Range: -17.625 dB (0x00) ~ 30 dB (0x7F). Default: 0x66 (~20 dB).

This can be changed at runtime using CMD_AUDIO_SET_ADC_GAIN.

DAC_gain

DAC digital volume controls the output (digital-to-analog) gain. Range: -65.625 dB (0x00) ~ 0 dB (0xAF). Default: 0xAF (0 dB).

This can be changed at runtime using CMD_AUDIO_SET_DAC_GAIN.

Note

Configure digital gain first. Only use analog gain (mic_gain) when digital gain alone is insufficient. Excessive analog gain introduces noticeable noise.

mix_mode

When set to 1, the module mixes up to 4 input PCM streams into a single output. This enables multi-stream mixing (e.g., local playback + network audio). Default: 0 (single input).

hpf_set

High-pass filter cutoff index (0–7). The cutoff frequency is approximately:

fc ≈ 5e-3 / (hpf_fc + 1) × fs

Default: 0 (HPF disabled). Use this to filter out DC offset and low-frequency noise.

enable_record

Set to 1 to enable the audio recording data path. Default: 0.

avsync_en

Set to 1 to enable hardware-aligned timestamps for A/V sync. Required when combining audio with video in a MIMO pipeline (MP4 recording, RTSP streaming).

Audio Module Configuration Timing

Configure audio_params_t before CMD_AUDIO_APPLY. After apply and stream start, keep runtime changes to commands that the audio control path can apply safely.

If a running stream must change the sample rate, mic type, channel count, or HPF setting, update audio_params_t and call CMD_AUDIO_SET_RESET to reinitialize the audio codec:

audio_params.sample_rate = ASR_48KHZ;
mm_module_ctrl(audio_ctx, CMD_AUDIO_SET_PARAMS, (int)&audio_params);
mm_module_ctrl(audio_ctx, CMD_AUDIO_SET_RESET, 0);  // reinitialize
mm_module_ctrl(audio_ctx, CMD_AUDIO_SET_TRX, 1);    // restart TX+RX

Common customer requests

Request	Supported timing	What to do
Change sample rate	Init-time only	Set `sample_rate` in `audio_params_t`, then call `CMD_AUDIO_SET_RESET` to reinitialize the codec.
Change mic type	Init-time only	Set `use_mic_type` in `audio_params_t`, then call `CMD_AUDIO_SET_RESET` to reinitialize the codec.
Tune mic volume	Runtime modify supported	Use `CMD_AUDIO_SET_ADC_GAIN`, value 0x00–0x7F (−17.625 dB to 30 dB).
Tune speaker volume	Runtime modify supported	Use `CMD_AUDIO_SET_DAC_GAIN`, value 0x00–0xAF (−65.625 dB to 0 dB).
Mute / unmute mic	Runtime operation supported	Use `CMD_AUDIO_SET_MIC_ENABLE` (0 = mute, 1 = unmute, zeros the data path).
Mute / unmute speaker	Runtime operation supported	Use `CMD_AUDIO_SET_SPK_ENABLE` (0 = mute, 1 = unmute, zeros the data path).
Start / stop capture	Runtime operation supported	Use `CMD_AUDIO_SET_RX` (0 = stop, 1 = start).
Start / stop playback	Runtime operation supported	Use `CMD_AUDIO_SET_TX` (0 = stop, 1 = start).
Enable / disable AEC	Runtime operation supported	Use `CMD_AUDIO_RUN_AEC` (0 = disable, 1 = enable). Dynamically toggles AEC processing.
Change AEC level	Runtime modify supported	Use `CMD_AUDIO_SET_AEC_LEVEL` (1–50, higher = more aggressive).
Enable / disable NS	Runtime operation supported	Use `CMD_AUDIO_RUN_NS` (0 = disable, 1–3 = noise suppression level).
Enable / disable AGC	Runtime operation supported	Use `CMD_AUDIO_RUN_AGC` (0 = disable, 1–3 = automatic gain control level).
Update EQ coefficients	Runtime modify supported	Update the EQ fields in `audio_params_t`, then call `CMD_AUDIO_MICEQ_RESET` or `CMD_AUDIO_SPKEQ_RESET` to apply.
Change HPF cutoff	Init-time only	Set `hpf_set` in `audio_params_t`, then call `CMD_AUDIO_SET_RESET` to reinitialize the codec.

CMD_AUDIO_* Parameter Reference

The tables below map every audio parameter to its configuration timing and the exact AP-side command to use.

Init-time only — configure in audio_params_t before CMD_AUDIO_APPLY. Changing these at runtime requires calling CMD_AUDIO_SET_RESET.

Parameter	`audio_params_t` field	Description
Sample rate	`sample_rate`	`ASR_8KHZ` / `ASR_16KHZ` / `ASR_32KHZ` / `ASR_44p1KHZ` / `ASR_48KHZ` / `ASR_88p2KHZ` / `ASR_96KHZ`
Word length	`word_length`	`WL_16BIT` or `WL_24BIT`
Channel count	`channel`	1 (mono) or 2 (stereo)
Mic type	`use_mic_type`	`USE_AUDIO_AMIC` / `USE_AUDIO_LEFT_DMIC` / `USE_AUDIO_RIGHT_DMIC` / `USE_AUDIO_STEREO_DMIC`
AMIC analog gain	`mic_gain`	`MIC_0DB` / `MIC_20DB` / `MIC_30DB` / `MIC_40DB`
DMIC left boost	`dmic_l_gain`	`DMIC_BOOST_0DB` / `DMIC_BOOST_12DB` / `DMIC_BOOST_24DB` / `DMIC_BOOST_36DB`
DMIC right boost	`dmic_r_gain`	Same options as left DMIC
HPF cutoff	`hpf_set`	0–7. fc ≈ 5e-3 / (hpf_fc + 1) × fs
Mix mode	`mix_mode`	0 = single input, 1 = mix up to 4 inputs
Recording path	`enable_record`	0 = disabled, 1 = enabled
A/V sync	`avsync_en`	0 = disabled, 1 = enabled (for MP4 / RTSP)
Mic EQ (5 biquads)	`mic_l_eq[5]` / `mic_r_eq[5]`	Per-biquad enable + coefficients (see EQ Setting)
Speaker EQ (5 biquads)	`spk_l_eq[5]`	Per-biquad enable + coefficients (see EQ Setting)

Init + Runtime — may be set in audio_params_t before CMD_AUDIO_APPLY as the initial value, and updated after stream start using the listed command.

Init field	Runtime CMD	API function	Description
`ADC_gain`	`CMD_AUDIO_SET_ADC_GAIN`	`audio_adc_digital_vol()`	ADC digital volume (−17.625 to 30 dB)
`DAC_gain`	`CMD_AUDIO_SET_DAC_GAIN`	`audio_dac_digital_vol()`	DAC digital volume (−65.625 to 0 dB)
RX ASP params	`CMD_AUDIO_SET_RXASP_PARAM`	—	Set RX ASP configuration (all AEC/AGC/NS settings)
TX ASP params	`CMD_AUDIO_SET_TXASP_PARAM`	—	Set TX ASP configuration (all AGC/NS settings)
Mic EQ	`CMD_AUDIO_MICEQ_RESET`	`audio_input_l_eq()` / `r_eq()`	Reapply mic EQ from current `audio_params_t`
Speaker EQ	`CMD_AUDIO_SPKEQ_RESET`	`audio_output_l_eq()`	Reapply speaker EQ from current `audio_params_t`

Runtime only — available only after CMD_AUDIO_APPLY. These have no init-time equivalent in audio_params_t.

CMD	Arg / usage	Description
`CMD_AUDIO_SET_MIC_ENABLE`	0 = mute, 1 = unmute	Mute/unmute microphone input (zeros the data path)
`CMD_AUDIO_SET_SPK_ENABLE`	0 = mute, 1 = unmute	Mute/unmute speaker output (zeros the data path)
`CMD_AUDIO_SET_TX`	0 = stop, 1 = start	Start / stop TX (playback) without affecting RX
`CMD_AUDIO_SET_RX`	0 = stop, 1 = start	Start / stop RX (capture) without affecting TX
`CMD_AUDIO_SET_TRX`	0 = stop, 1 = start	Start / stop both TX and RX simultaneously
`CMD_AUDIO_RUN_AEC`	0 = disable, 1 = enable	Dynamically enable / disable AEC processing
`CMD_AUDIO_RUN_NS`	0 = disable, 1–3 = aggressiveness	Dynamically enable / disable NS processing
`CMD_AUDIO_RUN_AGC`	0 = disable, 1–3 = aggressiveness	Dynamically enable / disable AGC processing
`CMD_AUDIO_RUN_VAD`	0 = disable, 1 = enable	Dynamically enable / disable voice activity detection
`CMD_AUDIO_SET_AEC_LEVEL`	1–50	Set AEC cancellation strength (higher = more aggressive)
`CMD_AUDIO_SET_AEC_ENABLE`	0 = disable, 1 = enable	Enable / disable AEC init at next reset
`CMD_AUDIO_SET_NS_ENABLE`	0–3	Enable / disable NS init at next reset
`CMD_AUDIO_SET_AGC_ENABLE`	0–3	Enable / disable AGC init at next reset
`CMD_AUDIO_SET_AEC_MODE`	AEC mode value	Set AEC mode (NEWAEC only)
`CMD_AUDIO_SET_MESSAGE_LEVEL`	0–3	Set audio debug log level (0 = none, 1 = all, 2 = warn+err, 3 = err only)
`CMD_AUDIO_SET_RESET`	0	Reinitialize audio codec with current `audio_params_t`
`CMD_AUDIO_SET_SAMPLERATE`	ASR constant	Update sample rate field in params (requires `CMD_AUDIO_SET_RESET`)
`CMD_AUDIO_GET_FRAMESIZE_MS`	Returns int (via pointer)	Get audio frame duration in milliseconds
`CMD_AUDIO_GET_FIRST_DATA_TS`	Returns uint32_t (via pointer)	Get timestamp of first data frame (useful for A/V sync)
`CMD_AUDIO_FORCE_DEINIT`	0	Force deinitialize audio codec

Audio Codec

Audio encoding and decoding are provided by standalone MMF modules documented in the MMF Development Guide :

aac_module — AAC encoder (PCM to AAC bitstream)
aad_module — AAC decoder (AAC bitstream to PCM)

Bitrate Selection Guide

Sample Rate	Bitrate	Use Case
8 kHz	16 kbps	Voice, narrowband
16 kHz	24–32 kbps	Voice, wideband
24 kHz	48 kbps	Good quality audio
48 kHz	64–128 kbps	High quality stereo

Note

Higher bitrates provide better quality but require more bandwidth. For real-time streaming applications, consider network conditions when selecting bitrate.

Audio Optimization

This section describes the software and hardware optimization solutions for audio.

Audio Setting

Gain Setting

Analog Microphone Gain Setting

The audio analog input gain can be divided into analog gain and digital gain.

Analog Mic Gain

It supports 0, 20, 30, 40 dB for gain optimization.

User can use audio_mic_analog_gain or set the parameter mic_gain for audio module to set it.

ADC Gain - ADC Volume

The ADC gain can be used to set the input (analog to digital) gain.

The range is -17.625dB (0x00) ~ 30dB (0x7F).

User can use the function audio_adc_digital_vol or use CMD_AUDIO_SET_ADC_GAIN to control the audio module.

A digital gain configuration is offered to control the audio output gain. Customers can set a reasonable gain value via DAC Volume to obtain the appropriate audio output volume. Basically setting the gain to 0dB (0xAF), the output amplitude will meet the board audio output volume requirements. Note that sound breakage will happen when the output gain is set too large.

If the analog gain is too large, it will affect the sound effect and noise will be obvious.

Recommendation: Customers should first configure the digital gain. If the audio signal gain needs to increase but the digital gain achieves the maximum range, then configure the analog gain.

Digital Microphone Gain Setting

Microphone

Description

Left Mic Gain

Left DMIC gain supports 0, 12, 24, 36 dB

for gain optimization. Use audio_l_dmic_gain or set dmic_l_gain parameter.

Right Mic Gain

Right DMIC gain supports 0, 12, 24, 36 dB

for gain optimization. Use audio_r_dmic_gain or set dmic_r_gain parameter.

DAC Gain - DAC Volume

The DAC gain can be used to set the output digital (to analog) gain.

The range is -65.625dB (0x00) ~ 0dB (0xAF).

User can use the function audio_dac_digital_vol or use CMD_AUDIO_SET_DAC_GAIN to control the audio module.

HPF Setting

A high pass filter is provided for users to filter low frequency noise. This is used to filter out noise from DC power; it is suggested to set the default value 0. If users want to use other filters, please refer to EQ setting.

Here is the function:

void audio_adc_l_hpf(audio_t *obj, BOOL en, audio_hpf_fc hpf_fc);

The parameters mean:

Parameter	Description
obj	Audio object defined in application software
en	Enable the high pass filter or not
hpf_fc	Set the cutoff frequency, value is 0~7; fc ~= 5e-3 / (hpf_fc + 1) * fs

EQ Setting

Users can use five sets of biquad filters in three sides for left digital mic (analog mic), right digital mic and audio output.

One biquad filter can switch to high-pass, low-pass, band-pass, notch, peak, low shelf, and high shelf filter by register settings.

Here are some tips for users to use the EQ:

Select the Biquad Filter

Users can use the following websites to configure the preferred filter type, sample rate, cutoff frequency, Q value and Gain first:

https://www.earlevel.com/main/2021/09/02/biquad-calculator-v3/

Get the Register Values

Users can use AmebaPro3_EQ_tool.exe to generate register settings. For example, if we choose a high pass filter with cutoff frequency 200Hz and Q value 0.707, users can type the setting and get the register values (0x1e45618, 0x1c000000, 0x2000000, 0x3c72d61, 0x1e35d500) for this setting.

Set the Register Value

After getting the register values, users can use the following functions to apply the filter setting:

void audio_input_l_eq(audio_t *obj, audio_eq eq, BOOL en, u32 h0, u32 b0, u32 b1, u32 a0, u32 a1);
void audio_input_r_eq(audio_t *obj, audio_eq eq, BOOL en, u32 h0, u32 b0, u32 b1, u32 a0, u32 a1);
void audio_output_l_eq(audio_t *obj, audio_eq eq, BOOL en, u32 h0, u32 b0, u32 b1, u32 a0, u32 a1);

Here are the parameters:

Parameter	Description
obj	Audio object defined in application software
eq	Select the EQ number, can be 0~4
en	Enable the EQ filter or not
h0, b0, b1, a0, a1	The register values obtained from AmebaPro3_EQ_tool.exe

Users can also set the EQ parameters in audio_params_t when using MMF settings.

Other Settings

Command	Description
CMD_AUDIO_SET_RESET	Re-initialize audio setting and ASP algorithms. Reset audio when configuration changes
CMD_AUDIO_SET_SAMPLERATE	Set sample rate. Reset required to apply
CMD_AUDIO_SET_TRX	Stop/start TX and RX without re-initializing
CMD_AUDIO_SET_MIC_ENABLE	Mute/unmute microphone input (sets data to 0)
CMD_AUDIO_SET_SPK_ENABLE	Mute/unmute speaker output (sets data to 0)

Note

If using audio codec, be sure the sample rate is fitting the sample rate used in audio codec.

Audio ASP Algorithm

The following table shows some common audio problems with their causes and also the adjustment using ASP algorithm.

Situation	Algorithm	Cases
Distortion	AGC	The ambient sound is too high Headphone preGain
Low audio volume	AGC	The original input volume is too low AGC is not working properly
Echo or howling	AEC	Too close between transmitting and receiving Volume too large or mic too sensitive AEC is not turned on AEC parameters is not set correctly
Intermittent voice	AEC, NS	NS or AEC suppression
Noise floor	NS	NS mode setting too low Caused by environment, NS can’t do well
Mechanical sound	Network, Device	Poor network environment Device sampling unstable or hardware issue

Note

The audio signal processing (ASP) is based on the digital audio signal. If the audio signal has already has the distortion, the ASP has no promise to get the expected result.

Enable ASP Algorithm

For using ASP algorithm, user needs to enable the ASP library in the build configuration.

Enable ENABLE_ASP in module_audio.h and use the 3A (AGC: Automatic gain control; ANS: Adaptive noise suppression; AEC: Acoustic echo cancellation) algorithms to obtain better audio effects.

Note

The parameters, sample_rate and mic_gain, and the initialization of NS, AEC, AGC and other algorithms will be set at CMD_AUDIO_APPLY and CMD_AUDIO_SET_RESET.

To enable ASP function, users can use the following parameters in ASP.h:

// =================== Open ASP algorithm (ASP.h) ================
typedef struct CTNS_cfg_s {
    int16_t NS_EN;
    int NSLevel;
    int16_t HPFEnable;
    int16_t QuickConvergenceEnable;
    int16_t Reserve1;
} CTNS_cfg_t;

typedef struct CTAGC_cfg_s {
    int16_t AGC_EN;
    CT_AGC_MODE AGCMode;
    int16_t ReferenceLvl;
    int16_t RatioFormat;    // Ratio format: 0 => integer, range 1~50, 1 => 8.8 fix point, range 26~50*256 (mapping 26/256~50)
    int16_t AttackTime;
    int16_t ReleaseTime;
    int16_t Ratio[3];
    int16_t Threshold[3];   // Threshold1, Threshold2, NoiseGateLvl
    int16_t KneeWidth;
    int16_t NoiseFloorAdaptEnable;
    int16_t RMSDetectorEnable;
    int16_t MaxGainLimit;
} CTAGC_cfg_t;

typedef struct CTAEC_cfg_s {
    int16_t AEC_EN;
    int16_t EchoTailLen;
    int16_t CNGEnable;
    int16_t PPLevel;
    int16_t DTControl;
    int16_t ConvergenceTime;
    int16_t Reserve1;
} CTAEC_cfg_t;

typedef struct VQE_SND_STATE_s {
    int16_t DoA;              // in degrees
    int16_t ERLE;             // in dB
    int16_t SinLvldB;         // in dBFs
    int16_t SoutLvldB;        // in dBFs after AGC (if AGC is enabled)
    int16_t DTState;          // 0 = single talk or 1 = double talk
    int16_t HCDetectState;    // 1 = detected, 0 = not detected
    uint8_t AECRun;
    uint8_t AGCRun;
    uint8_t NSRun;
    uint8_t BFRun;
    uint8_t Reserve1;
    uint8_t Reserve2;
    uint8_t Reserve3;
    uint8_t Reserve4;
} VQE_SND_STATE_t;

typedef struct VQE_RCV_STATE_s {
    int16_t RinLvldB;
    int16_t RoutLvldB;
    int16_t HCDetectState;    // 1 = detected, 0 = not detected
    uint8_t AGCRun;
    uint8_t NSRun;
    uint8_t Reserve1;
    uint8_t Reserve2;
    uint8_t Reserve3;
    uint8_t Reserve4;
} VQE_RCV_STATE_t;

Parameter Details

CTAEC_cfg_t (AEC Configuration)

Parameter	Description
AEC_EN	Enable the AEC module in AEC process
EchoTailLen	Buffer length for echo cancel; higher values increase CPU usage. Suggest 64 for 16KHz, 128 for 8KHz. Support 32/64/128
CNGEnable	Enable comfort noise generation (0 or 1)
PPLevel	AEC fine tune; higher = more aggressive echo cancel (may cancel more local). Support 1~50
DTControl	AEC coarse tune: 1 (allow some residual), 2 (attenuate up to 6dB local), 3 (attenuate up to 9dB local)
ConvergenceTime	AEC initialization convergence time in msec, support 100~1000

CTAGC_cfg_t (AGC Configuration)

Parameter	Description
AGC_EN	Enable the AGC module in the AGC process
AGCMode	AGC mode: 0 (CT_ALC), 1 (CT_LIMITER)
ReferenceLvl	Output target reference level (dBFS), support 0,1,…,30 (0,-1,…,-30dBFs)
RatioFormat	Ratio format: 0 = integer (1~50), 1 = 8.8 fix point (26~50*256)
AttackTime	Signal amplitude compression transition time (ms), support 1~500
ReleaseTime	Signal amplitude boost transition time (ms), support 1~500
Ratio[3]	Three ratios for adjusting AGC gain curve
Threshold[3]	Three thresholds: Threshold1, Threshold2 (0~81), NoiseGateLvl (50~90)
KneeWidth	AGC gain curve soft knee width, support 0~10
NoiseFloorAdaptEnable	Enable noise detect on AGC (ignore background noise), 0 or 1
RMSDetectorEnable	0 = peak detection, 1 = RMS detection
MaxGainLimit	Maximum gain in dB for AGC, support 6,12,18,24,30

CTNS_cfg_t (NS Configuration)

Parameter	Description
NS_EN	Enable the NS module in the NS process
NSLevel	NS aggressiveness in dB (higher = more aggressive), support 3~35
HPFEnable	Enable HPF before NS, 0 or 1
QuickConvergenceEnable	NS convergence speed: 1 = immediately suppress (quick), 0 = smooth suppress

ASP Algorithm Usage

Here are the configurations for ASP algorithm:

Configuration	Description
Supported Sample Rates	8K and 16K audio
Default Settings	Defined in module_audio.c as `default_rx_asp_params` and `default_tx_asp_params`
Get RX ASP Parameters	Use `CMD_AUDIO_GET_RXASP_PARAM`
Get TX ASP Parameters	Use `CMD_AUDIO_GET_TXASP_PARAM`
Set RX ASP Parameters	Use `CMD_AUDIO_SET_RXASP_PARAM`
Set TX ASP Parameters	Use `CMD_AUDIO_SET_TXASP_PARAM`
AGC Enable (RX/TX)	Set `agc_cfg.AGC_EN` to 1 to enable, 0 to disable
NS Enable (RX/TX)	Set `ns_cfg.NS_EN` to 1 to enable, 0 to disable
AEC Enable (RX)	Set `aec_cfg.AEC_EN` to 1 to enable, 0 to disable

AEC Setting

The AEC algorithm includes three parts: delay adjustment strategy, linear echo estimation, and nonlinear echo suppression.

Command	Description
CMD_AUDIO_RUN_AEC	Dynamically switch AEC_process() usage
CMD_AUDIO_SET_AEC_ENABLE	Enable/disable AEC_init() during audio reset
CMD_AUDIO_SET_AEC_LEVEL	Set echo cancellation strength

NS Setting

The NS algorithm is aimed at decreasing the noise or environment sound, so it is recommended to use before other ASP algorithms.

Command	Description
CMD_AUDIO_SET_NS_ENABLE	Enable/disable NSx_init() during audio reset
CMD_AUDIO_RUN_NS	Dynamically switch NSx_process() usage

AGC Setting

The AGC algorithm is used to balance the audio volume of signal streaming.

Command	Description
CMD_AUDIO_SET_AGC_ENABLE	Enable/disable AGC_init() during audio reset
CMD_AUDIO_RUN_AGC	Dynamically switch AGC_process() usage

For media example usage, see Media Example .