Hass/xiaozhi-esp32-server
1
0
Fork 0
mirror of https://github.com/xinnan-tech/xiaozhi-esp32-server.git synced 2026-07-28 19:03:51 +08:00
Code Issues Packages Projects Releases Wiki Activity

mqtt

Browse Source
This commit is contained in:
FAN-yeB
2025-09-08 11:04:42 +08:00
parent d04ec9d510
commit eed9503391
11 changed files with 290 additions and 28 deletions
Download Patch File Download Diff File Expand all files Collapse all files
main/xiaozhi-server/core/connection.py
+82
View File
@@ -281,8 +281,86 @@ class ConnectionHandler:
return
if self.asr is None:
return
if len(message) >= 16:
try:
timestamp = int.from_bytes(message[8:12], 'big')
audio_length = int.from_bytes(message[12:16], 'big')
# 提取音频数据
if audio_length > 0 and len(message) >= 16 + audio_length:
audio_data = message[16:16 + audio_length]
# 基于时间戳进行简单排序
self._process_websocket_audio(audio_data, timestamp)
return
elif len(message) > 16:
# 去掉16字节头部
audio_data = message[16:]
self.asr_audio_queue.put(audio_data)
return
except Exception as e:
self.logger.bind(tag=TAG).error(f"解析WebSocket音频包失败: {e}")
self.asr_audio_queue.put(message)
def _process_websocket_audio(self, audio_data, timestamp):
"""处理WebSocket格式的音频包"""
# 初始化时间戳序列管理
if not hasattr(self, 'audio_timestamp_buffer'):
self.audio_timestamp_buffer = {}
self.last_processed_timestamp = 0
self.max_timestamp_buffer_size = 20
# 如果时间戳是递增的，直接处理
if timestamp >= self.last_processed_timestamp:
self.asr_audio_queue.put(audio_data)
self.last_processed_timestamp = timestamp
# 处理缓冲区中的后续包
processed_any = True
while processed_any:
processed_any = False
for ts in sorted(self.audio_timestamp_buffer.keys()):
if ts > self.last_processed_timestamp:
buffered_audio = self.audio_timestamp_buffer.pop(ts)
self.asr_audio_queue.put(buffered_audio)
self.last_processed_timestamp = ts
processed_any = True
break
else:
# 乱序包，暂存
if len(self.audio_timestamp_buffer) < self.max_timestamp_buffer_size:
self.audio_timestamp_buffer[timestamp] = audio_data
else:
self.asr_audio_queue.put(audio_data)
def _process_sequenced_audio(self, audio_data, sequence, timestamp):
"""处理有序的音频包"""
# 初始化音频缓冲区
if not hasattr(self, 'audio_buffer'):
self.audio_buffer = {}
self.expected_sequence = sequence
self.max_buffer_size = 20 # 最大缓冲20个包
# 如果是下一个期望的包，直接处理
if sequence == self.expected_sequence:
self.asr_audio_queue.put(audio_data)
self.expected_sequence += 1
# 检查缓冲区中是否有后续的连续包
while self.expected_sequence in self.audio_buffer:
buffered_audio = self.audio_buffer.pop(self.expected_sequence)
self.asr_audio_queue.put(buffered_audio)
self.expected_sequence += 1
elif sequence > self.expected_sequence:
# 乱序包，暂存到缓冲区
if len(self.audio_buffer) < self.max_buffer_size:
self.audio_buffer[sequence] = audio_data
async def handle_restart(self, message):
"""处理服务器重启请求"""
try:
@@ -921,6 +999,10 @@ class ConnectionHandler:
async def close(self, ws=None):
"""资源清理方法"""
try:
# 清理音频缓冲区
if hasattr(self, 'audio_buffer'):
self.audio_buffer.clear()
# 取消超时任务
if self.timeout_task and not self.timeout_task.done():
self.timeout_task.cancel()
main/xiaozhi-server/core/handle/receiveAudioHandle.py
+17 -17
View File
@@ -1,11 +1,12 @@
import time
import json
import asyncio
from core.utils.util import audio_to_data
from core.handle.abortHandle import handleAbortMessage
from core.handle.sendAudioHandle import send_stt_message
from core.handle.intentHandler import handle_user_intent
from core.utils.output_counter import check_device_output_limit
from core.handle.sendAudioHandle import send_stt_message, SentenceType
from core.handle.abortHandle import handleAbortMessage
import time
import asyncio
import json
from core.handle.sendAudioHandle import SentenceType
from core.utils.util import audio_to_data
TAG = __name__
@@ -21,6 +22,7 @@ async def handleAudioMessage(conn, audio):
if not hasattr(conn, "vad_resume_task") or conn.vad_resume_task.done():
conn.vad_resume_task = asyncio.create_task(resume_vad_detection(conn))
return
if have_voice:
if conn.client_is_speaking:
await handleAbortMessage(conn)
@@ -29,16 +31,18 @@ async def handleAudioMessage(conn, audio):
# 接收音频
await conn.asr.receive_audio(conn, audio, have_voice)
async def resume_vad_detection(conn):
# 等待2秒后恢复VAD检测
await asyncio.sleep(1)
conn.just_woken_up = False
async def startToChat(conn, text):
# 检查输入是否是JSON格式（包含说话人信息）
speaker_name = None
actual_text = text
try:
# 尝试解析JSON格式的输入
if text.strip().startswith('{') and text.strip().endswith('}'):
@@ -47,13 +51,13 @@ async def startToChat(conn, text):
speaker_name = data['speaker']
actual_text = data['content']
conn.logger.bind(tag=TAG).info(f"解析到说话人信息: {speaker_name}")
# 直接使用JSON格式的文本，不解析
actual_text = text
except (json.JSONDecodeError, KeyError):
# 如果解析失败，继续使用原始文本
pass
# 保存说话人信息到连接对象
if speaker_name:
conn.current_speaker = speaker_name
@@ -114,12 +118,10 @@ async def no_voice_close_connect(conn, have_voice):
async def max_out_size(conn):
# 播放超出最大输出字数的提示
conn.client_abort = False
text = "不好意思，我现在有点事情要忙，明天这个时候我们再聊，约好了哦！明天不见不散，拜拜！"
await send_stt_message(conn, text)
file_path = "config/assets/max_output_size.wav"
opus_packets = audio_to_data(file_path)
opus_packets, _ = audio_to_data(file_path)
conn.tts.tts_audio_queue.put((SentenceType.LAST, opus_packets, text))
conn.close_after_chat = True
@@ -138,7 +140,7 @@ async def check_bind_device(conn):
# 播放提示音
music_path = "config/assets/bind_code.wav"
opus_packets = audio_to_data(music_path)
opus_packets, _ = audio_to_data(music_path)
conn.tts.tts_audio_queue.put((SentenceType.FIRST, opus_packets, text))
# 逐个播放数字
@@ -146,17 +148,15 @@ async def check_bind_device(conn):
try:
digit = conn.bind_code[i]
num_path = f"config/assets/bind_code/{digit}.wav"
num_packets = audio_to_data(num_path)
num_packets, _ = audio_to_data(num_path)
conn.tts.tts_audio_queue.put((SentenceType.MIDDLE, num_packets, None))
except Exception as e:
conn.logger.bind(tag=TAG).error(f"播放数字音频失败: {e}")
continue
conn.tts.tts_audio_queue.put((SentenceType.LAST, [], None))
else:
# 播放未绑定提示
conn.client_abort = False
text = f"没有找到该设备的版本信息，请正确配置 OTA地址，然后重新编译固件。"
await send_stt_message(conn, text)
music_path = "config/assets/bind_not_found.wav"
opus_packets = audio_to_data(music_path)
opus_packets, _ = audio_to_data(music_path)
conn.tts.tts_audio_queue.put((SentenceType.LAST, opus_packets, text))
main/xiaozhi-server/core/handle/sendAudioHandle.py
+42 -8
View File
@@ -55,6 +55,7 @@ async def sendAudio(conn, audios, frame_duration=60):
"last_send_time": 0,
"packet_count": 0,
"start_time": time.perf_counter(),
"sequence": 0, # 添加序列号
}
flow_control = conn.audio_flow_control
@@ -67,11 +68,22 @@ async def sendAudio(conn, audios, frame_duration=60):
if delay > 0:
await asyncio.sleep(delay)
# 发送数据包
await conn.websocket.send(audios)
# 为opus数据包添加16字节头部
timestamp = int((flow_control["start_time"] + flow_control["packet_count"] * frame_duration / 1000) * 1000) % (2**32)
header = bytearray(16)
header[0] = 1 # type
header[2:4] = len(audios).to_bytes(2, 'big') # payload length
header[4:8] = flow_control["sequence"].to_bytes(4, 'big') # connection id/sequence
header[8:12] = timestamp.to_bytes(4, 'big') # 时间戳
header[12:16] = len(audios).to_bytes(4, 'big') # opus长度
# 发送包含头部的完整数据包
complete_packet = bytes(header) + audios
await conn.websocket.send(complete_packet)
# 更新流控状态
flow_control["packet_count"] += 1
flow_control["sequence"] += 1
flow_control["last_send_time"] = time.perf_counter()
else:
# 文件型音频走普通播放
@@ -81,11 +93,21 @@ async def sendAudio(conn, audios, frame_duration=60):
# 执行预缓冲
pre_buffer_frames = min(3, len(audios))
for i in range(pre_buffer_frames):
await conn.websocket.send(audios[i])
# 为预缓冲包添加头部
timestamp = int((start_time + i * frame_duration / 1000) * 1000) % (2**32)
header = bytearray(16)
header[0] = 1 # type
header[2:4] = len(audios[i]).to_bytes(2, 'big') # payload length
header[4:8] = i.to_bytes(4, 'big') # sequence
header[8:12] = timestamp.to_bytes(4, 'big') # 时间戳
header[12:16] = len(audios[i]).to_bytes(4, 'big') # opus长度
complete_packet = bytes(header) + audios[i]
await conn.websocket.send(complete_packet)
remaining_audios = audios[pre_buffer_frames:]
# 播放剩余音频帧
for opus_packet in remaining_audios:
for i, opus_packet in enumerate(remaining_audios):
if conn.client_abort:
break
@@ -98,9 +120,21 @@ async def sendAudio(conn, audios, frame_duration=60):
delay = expected_time - current_time
if delay > 0:
await asyncio.sleep(delay)
await conn.websocket.send(opus_packet)
# 为opus数据包添加16字节头部 (timestamp at offset 8, length at offset 12)
timestamp = int((start_time + play_position / 1000) * 1000) % (2**32) # 使用播放位置计算时间戳
sequence = pre_buffer_frames + i # 确保序列号连续
header = bytearray(16)
header[0] = 1 # type
header[2:4] = len(opus_packet).to_bytes(2, 'big') # payload length
header[4:8] = sequence.to_bytes(4, 'big') # sequence
header[8:12] = timestamp.to_bytes(4, 'big') # 时间戳在第8-11字节
header[12:16] = len(opus_packet).to_bytes(4, 'big') # opus长度在第12-15字节
# 发送包含头部的完整数据包
complete_packet = bytes(header) + opus_packet
await conn.websocket.send(complete_packet)
play_position += frame_duration
main/xiaozhi-server/core/providers/tts/base.py
+2 -1
View File
@@ -336,7 +336,8 @@ class TTSProviderBase(ABC):
self.conn.audio_flow_control = {
'last_send_time': 0,
'packet_count': 0,
'start_time': time.perf_counter()
'start_time': time.perf_counter(),
'sequence': 0 # 添加序列号
}
# 上报TTS数据