mirror of
https://github.com/xinnan-tech/xiaozhi-esp32-server.git
synced 2026-07-25 00:23:53 +08:00
135 lines
4.0 KiB
Python
135 lines
4.0 KiB
Python
from config.logger import setup_logging
|
|
import json
|
|
import asyncio
|
|
import time
|
|
from core.utils.util import (
|
|
get_string_no_punctuation_or_emoji,
|
|
analyze_emotion
|
|
)
|
|
|
|
TAG = __name__
|
|
logger = setup_logging()
|
|
|
|
emoji_map = {
|
|
'neutral': '😶',
|
|
'happy': '🙂',
|
|
'laughing': '😆',
|
|
'funny': '😂',
|
|
'sad': '😔',
|
|
'angry': '😠',
|
|
'crying': '😭',
|
|
'loving': '😍',
|
|
'embarrassed': '😳',
|
|
'surprised': '😲',
|
|
'shocked': '😱',
|
|
'thinking': '🤔',
|
|
'winking': '😉',
|
|
'cool': '😎',
|
|
'relaxed': '😌',
|
|
'delicious': '🤤',
|
|
'kissy': '😘',
|
|
'confident': '😏',
|
|
'sleepy': '😴',
|
|
'silly': '😜',
|
|
'confused': '🙄'
|
|
}
|
|
|
|
async def sendAudioMessage(conn, audios, text, text_index=0):
|
|
# 发送句子开始消息
|
|
emotion = analyze_emotion(text)
|
|
emoji = emoji_map.get(emotion, '🙂') # 默认使用笑脸
|
|
await conn.websocket.send(json.dumps(
|
|
{
|
|
"type": "llm",
|
|
"text": emoji,
|
|
"emotion": emotion,
|
|
"session_id": conn.session_id,
|
|
}
|
|
)
|
|
)
|
|
if text_index == conn.tts_first_text_index:
|
|
logger.bind(tag=TAG).info(f"发送第一段语音: {text}")
|
|
await send_tts_message(conn, "sentence_start", text)
|
|
|
|
is_first_audio = (text_index == conn.tts_first_text_index)
|
|
await sendAudio(conn, audios, pre_buffer=is_first_audio)
|
|
|
|
await send_tts_message(conn, "sentence_end", text)
|
|
|
|
# 发送结束消息（如果是最后一个文本）
|
|
if conn.llm_finish_task and text_index == conn.tts_last_text_index:
|
|
await send_tts_message(conn, "stop", None)
|
|
if conn.close_after_chat:
|
|
await conn.close()
|
|
|
|
|
|
# 播放音频
|
|
async def sendAudio(conn, audios, pre_buffer=True):
|
|
# 流控参数优化
|
|
frame_duration = 60 # 帧时长（毫秒），匹配 Opus 编码
|
|
start_time = time.perf_counter()
|
|
play_position = 0
|
|
last_reset_time = time.perf_counter() # 记录最后的重置时间
|
|
|
|
# 仅当第一句话时执行预缓冲
|
|
if pre_buffer:
|
|
pre_buffer_frames = min(3, len(audios))
|
|
for i in range(pre_buffer_frames):
|
|
await conn.websocket.send(audios[i])
|
|
remaining_audios = audios[pre_buffer_frames:]
|
|
else:
|
|
remaining_audios = audios
|
|
|
|
# 播放剩余音频帧
|
|
for opus_packet in remaining_audios:
|
|
if conn.client_abort:
|
|
return
|
|
|
|
# 每分钟重置一次计时器
|
|
if time.perf_counter() - last_reset_time > 60:
|
|
await conn.reset_timeout()
|
|
last_reset_time = time.perf_counter()
|
|
|
|
# 计算预期发送时间
|
|
expected_time = start_time + (play_position / 1000)
|
|
current_time = time.perf_counter()
|
|
delay = expected_time - current_time
|
|
if delay > 0:
|
|
await asyncio.sleep(delay)
|
|
|
|
await conn.websocket.send(opus_packet)
|
|
|
|
play_position += frame_duration
|
|
|
|
|
|
async def send_tts_message(conn, state, text=None):
|
|
"""发送 TTS 状态消息"""
|
|
message = {"type": "tts", "state": state, "session_id": conn.session_id}
|
|
if text is not None:
|
|
message["text"] = text
|
|
|
|
# TTS播放结束
|
|
if state == "stop":
|
|
# 播放提示音
|
|
tts_notify = conn.config.get("enable_stop_tts_notify", False)
|
|
if tts_notify:
|
|
stop_tts_notify_voice = conn.config.get(
|
|
"stop_tts_notify_voice", "config/assets/tts_notify.mp3"
|
|
)
|
|
audios, duration = conn.tts.audio_to_opus_data(stop_tts_notify_voice)
|
|
await sendAudio(conn, audios)
|
|
# 清除服务端讲话状态
|
|
conn.clearSpeakStatus()
|
|
|
|
# 发送消息到客户端
|
|
await conn.websocket.send(json.dumps(message))
|
|
|
|
|
|
async def send_stt_message(conn, text):
|
|
"""发送 STT 状态消息"""
|
|
stt_text = get_string_no_punctuation_or_emoji(text)
|
|
await conn.websocket.send(
|
|
json.dumps({"type": "stt", "text": stt_text, "session_id": conn.session_id})
|
|
)
|
|
await send_tts_message(conn, "start")
|