开发板：LicheeRV Nano / Duo-s
1、语音转文字ASR
将麦克风输入的语音实时转换为文字，并将语音识别的文字实时显示在LCD屏幕
阿里云API (阿里云登录 - 欢迎登录阿里云，安全稳定的云计算服务平台)
appKey = ‘s9NZm8ozBKyX63vK’
apt
token = ‘f1ce8db3539a4b08b775debdebc23a89’appKey = ‘P918jP30TLJNHi3Q’ #s9NZm8ozBKyX63vK’ #‘RxkHgzYYYYLIP4OD’
token = ‘080e17fafb8d4101ac25d5f778bd82fc’ #f1ce8db3539a4b08b775debdebc23a89’
2、CHAT
文字传入到大语言模型进行chat，将chat的response实时显示在LCD屏幕
大语言模型使用Kimi chat API
API_KEY = “sk-ycJsTdyonBbTzSSBnqogZUjjEUlhzZmEkKf1pZmsO0vd8FYA”
BASE_URL = “https://api.moonshot.cn/v1/chat/completions”，
智谱chat API
API_KEY = “ebb785194c713e7b419ca8742277d414.hCBC11QCZvC5N0YK”
BASE_URL = “https://open.bigmodel.cn/api/paas/v4/chat/completions”
3、语音合成TTS
将chat得到的response文字转换为语音在扬声器输出
host = ‘nls-gateway-cn-shanghai.aliyuncs.com’
4、语音唤醒
可执行文件编译步骤参考简介 | Milk-V
# 板子上操作
# 导入依赖库
export LD_LIBRARY_PATH='/mnt/system/lib'
# 运行语音唤醒
./sample_aud_order panns_cv181x.cvimodel 8000 2 0
LicheeRV nano
LicheeRV nano网络设置命令
# 在sd卡第一个分区创建wifi.sta文件启用sta模式:
touch /boot/wifi.sta
rm /boot/wifi.ap /boot/wifi.mon
# 然后将AP的SSID和密码写入文件:
echo ssid > /boot/wifi.ssid
echo pass > /boot/wifi.pass
# 重启Wifi服务
/etc/init.d/S30wifi stop
/etc/init.d/S30wifi start
亦可通过sh脚本一键执行：
vi wifi-nano.sh
# 执行i进入编辑模式，将以下内容写入
#######################################
#!/bin/bash
# 创建wifi.sta文件并启用sta模式
touch /boot/wifi.sta
rm /boot/wifi.ap /boot/wifi.mon
# 提示用户输入WiFi SSID
echo "Enter your WiFi SSID:"
read ssid_input
# 提示用户输入WiFi密码
echo "Enter your WiFi password:"
read -s pass_input
# 写入用户提供的SSID和密码到文件
echo "$ssid_input" > /boot/wifi.ssid
echo "$pass_input" > /boot/wifi.pass
# 重启Wifi服务
/etc/init.d/S30wifi stop
/etc/init.d/S30wifi start
#######################################
# ESC退出编辑模式，:wq保存退出
# 执行sh
sh wifi-nano.sh
LCD屏测试相关命令
# 清屏
cat /dev/zero > /dev/fb0
# 花屏
cat /dev/random > /dev/fb0
# 亦可测试lvgl的demo（显示lvgl官网样例）
麦克风及扬声器相关设置命令：
# 麦克风音量设置
amixer -Dhw:0 cset name='ADC Capture Volume' 24
# 录音命令(Ctrl+C结束录音)：
arecord -f dat -c 1 -r 16000 XXXX.wav
# 播放录音:
aplay XXXX.wav
# 扬声器播放音量设置（假设音量设置为24）
# 两种方式:
amixer -Dhw:3 cset name='Speaker Playback Volume' 24
amixer cset -c 3 numid=6 24
相关设备查看命令
# 查看录音设备
arecord -l
# 查看播放设备
aplay -l
# 查看具体设备号的信息（假设设备号为3）
amixer contents -c 3
##########################################################################
Duo-s
注：USB转TTL串口线序如下图，RX->TX TX->RX
Duo-s网络设置命令
# 编辑`/etc/wpa_supplicant.conf`文件，将内容更换为如下内容，
# 并替换 `ssid` 和 `psk` 为要连接的 WIFI 账号和密码
ctrl_interface=/var/run/wpa_supplicant
ap_scan=1
update_config=1 v
network={
ssid="wifi_test"
psk="12345678"
key_mgmt=WPA-PSK}
# 重启网络
wpa_supplicant -B -i wlan0 -c /etc/wpa_supplicant.conf
亦可通过sh脚本一键执行：
vi wifi-duo-s.sh
# 执行i进入编辑模式，将以下内容写入
#######################################
#!/bin/bash
# 提示用户输入WiFi的SSID和密码
read -p "请输入WiFi的SSID: " ssid
read -p "请输入WiFi的密码: " password
# 编辑 /etc/wpa_supplicant.conf 文件
cat <<EOF > /etc/wpa_supplicant.conf
ctrl_interface=/var/run/wpa_supplicant
ap_scan=1
update_config=1
network={
ssid="$ssid"
psk="$password"
key_mgmt=WPA-PSK
}
EOF
# 重启网络
wpa_supplicant -B -i wlan0 -c /etc/wpa_supplicant.conf
echo "WiFi configuration completed."
#######################################
# ESC退出编辑模式，:wq保存退出
# 执行sh
sh wifi-duo-s.sh
LCD屏测试相关命令
# 清屏
cat /dev/zero > /dev/fb0
# 花屏
cat /dev/random > /dev/fb0
# 亦可测试lvgl的demo（显示lvgl官网样例）
麦克风及扬声器相关设置命令：
==注：==使用USB转TTL串口、type-c供电方可使用声卡功能，需要使用专用版本的img
# 麦克风音量设置
amixer -Dhw:0 cset name='ADC Capture Volume' 24
# 录音命令(Ctrl+C结束录音)：
arecord -f dat -c 1 -r 16000 XXXX.wav
# 播放录音:
aplay XXXX.wav
#查看设备详情（假设设备号为3）
amixer contents -c 3
# 扬声器播放音量设置（假设音量设置为24）
# 两种方式:（假设设备号为3）
amixer -Dhw:3 cset name='Speaker Playback Volume' 24
amixer cset -c 3 numid=6 24
相关设备查看命令
# 查看录音设备
arecord -l
# 查看播放设备
aplay -l
# 查看具体设备号的信息（假设设备号为3）
amixer contents -c 3
语音识别-CHAT-语音合成asr_chat_tts整体代码：
0422版本–添加stream流式数据传输及更换智谱APIasr_chat-zp_tts_stream.py
# -*- coding: UTF-8 -*-
import http.client
import urllib.parse
import json
import subprocess
import time
import httpx
import requests
subprocess.Popen(['chmod', '+x', 'stable_demo'])
subprocess.Popen(['./stable_demo'])
print('Asr chat tts begin....')
appKey = 'P918jP30TLJNHi3Q'#s9NZm8ozBKyX63vK' #'RxkHgzYYYYLIP4OD'
token = '080e17fafb8d4101ac25d5f778bd82fc'
# Chat Configuration
API_KEY = "ebb785194c713e7b419ca8742277d414.hCBC11QCZvC5N0YK"
BASE_URL = "https://open.bigmodel.cn/api/paas/v4/chat/completions"
history = [{"role": "system", "content": "您好！"}]
# Aliyun
# url = 'https://nls-gateway-cn-shanghai.aliyuncs.com/stream/v1/asr'
host = 'nls-gateway-cn-shanghai.aliyuncs.com'
def record_on_gpio(pin):
is_pressed = False
audioFilepath = './output.wav'
while True:
try:
with open('/sys/class/gpio/gpio{}/value'.format(pin), 'r') as gpio_file:
value = gpio_file.read().strip()
#print('get key value {}',value)
if value == '1' and not is_pressed:
# 按键按下时开始录音
recording_process = subprocess.Popen(['arecord', '-f', 'dat', '-c', '1', '-r', '16000', 'output.wav'])
is_pressed = True
print("Recording started.")
if value == '0' and is_pressed:
subprocess.Popen(['killall', 'arecord'])
recording_process.wait() # 等待录音进程结束
is_pressed = False
print("Recording stopped.")
return audioFilepath
except Exception as e:
print("Error:", e)
def process_chunk(chunk,response_accumulator):
if chunk.strip() == "[DONE]":
return True, None
try:
data = json.loads(chunk)
# print('process_chunk data:', data)
if 'choices' in data and data['choices']:
for choice in data['choices']:
if 'delta' in choice and 'content' in choice['delta'] and choice['delta']['content']:
result = choice['delta']['content']
# print('process_chunk result:', result)
response_accumulator.append(result)
return False, result
except Exception as e:
print(f"处理数据块时出错: {e}")
return False, None
def chat(query, history):
history += [{"role": "user", "content": query}]
data = {
"model": "glm-4",
"messages": history,
"temperature": 0.3,
"stream": True,
}
headers = {
"Authorization": f"Bearer {API_KEY}",
"Content-Type": "application/json"
}
response_accumulator = []
response = requests.post(BASE_URL, data=json.dumps(data), headers=headers, stream=True)
send_to_lvgl(f"[CLEAR]{query}: ")
try:
for chunk in response.iter_lines():
if chunk:
chunk_str = chunk.decode("utf-8")
if chunk_str.startswith("data: "):
chunk_str = chunk_str[len("data: "):]
done, result = process_chunk(chunk_str,response_accumulator)
# print('result is', result)
chunk_str = "data: " + chunk_str
# print("Get response:", chunk_str)
if result:
send_to_lvgl(result)
if done:
tts_text = ''.join(response_accumulator)
tts_to_play(tts_text)
except Exception as e:
print(f"Error: {str(e)}")
def send_to_lvgl(text):
pipe_name = '/tmp/query_pipe'
try:
with open(pipe_name, 'w') as pipe:
pipe.write(text)
pipe.flush()
except Exception as e:
print(f"LVGL send error: {e}")
def process(request, token, audioFile) :
# 读取音频
print('process {} {}'.format(request, audioFile))
with open(audioFile, mode = 'rb') as f:
audioContent = f.read()
host = 'nls-gateway-cn-shanghai.aliyuncs.com'
# 设置HTTPS请求头部
httpHeaders = {
'X-NLS-Token': token,
'Content-type': 'application/octet-stream',
'Content-Length': len(audioContent)
}
conn = http.client.HTTPSConnection(host)
conn.request(method='POST', url=request, body=audioContent, headers=httpHeaders)
response = conn.getresponse()
print('Response status and response reason:')
print(response.status ,response.reason)
try:
body = json.loads(response.read())
text = body['result']
print('Recognized Text:', text)
chat_response = chat(text, history)
print('Chat Response:', chat_response)
except ValueError:
print('The response is not json format string')
conn.close()
def oneloop():
print('Wait for key press')
audioFilepath = record_on_gpio(499)
#print('Wait for first audio')
format = 'pcm'
sampleRate = 16000
enablePunctuationPrediction = True
enableInverseTextNormalization = True
enableVoiceDetection = False
# 设置RESTful请求参数
asrurl = f'https://{host}/stream/v1/asr'
request = asrurl + '?appkey=' + appKey
request = request + '&format=' + format
request = request + '&sample_rate=' + str(sampleRate)
if enablePunctuationPrediction :
request = request + '&enable_punctuation_prediction=' + 'true'
if enableInverseTextNormalization :
request = request + '&enable_inverse_text_normalization=' + 'true'
if enableVoiceDetection :
request = request + '&enable_voice_detection=' + 'true'
print('Request: ' + request)
process(request, token, audioFilepath)
def tts_to_play(text, file_path='response.wav'):
ttsurl = f'https://{host}/stream/v1/tts'
text_encoded = urllib.parse.quote_plus(text)
tts_request = f"{ttsurl}?appkey={appKey}&token={token}&text={text_encoded}&format=wav&sample_rate=16000"
conn = http.client.HTTPSConnection(host)
conn.request('GET', tts_request)
response = conn.getresponse()
body = response.read()
if response.status == 200 and response.getheader('Content-Type') == 'audio/mpeg':
with open(file_path, 'wb') as f:
f.write(body)
print('TTS audio saved successfully')
subprocess.Popen(['aplay', file_path])
else:
print('TTS request failed:', body)
conn.close()
while True:
try:
oneloop()
except Exception as e:
print(e)