mirror of
https://github.com/xinnan-tech/xiaozhi-esp32-server.git
synced 2026-07-21 22:53:56 +08:00
update:优化豆包流式ASR接口
This commit is contained in:
hrz
@@ -193,7 +193,7 @@ Websocket接口地址: wss://2662r3426b.vicp.fun/xiaozhi/v1/
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| 模块名称 | 入门全免费设置 | 全流式配置 |
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|---------|---------|------|
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| ASR(语音识别) | FunASR(本地) | ✅DoubaoASR(火山流式语音识别) |
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| ASR(语音识别) | FunASR(本地) | ✅DoubaoStreamASR(火山流式语音识别) |
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| LLM(大模型) | ChatGLMLLM(智谱glm-4-flash) | ✅DoubaoLLM(火山doubao-1-5-pro-32k-250115) |
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| VLLM(视觉大模型) | ChatGLMVLLM(智谱glm-4v-flash) | ✅QwenVLVLLM(千问qwen2.5-vl-3b-instructh) |
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| TTS(语音合成) | EdgeTTS(微软语音) | ✅HuoshanDoubleStreamTTS(火山双流式语音合成) |
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@@ -278,6 +278,7 @@ ASR:
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# 可以在这里申请相关Key等信息
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# https://console.volcengine.com/speech/app
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# DoubaoASR和DoubaoStreamASR的区别是：DoubaoASR是按次收费，DoubaoStreamASR是按时收费
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# 开通地址https://console.volcengine.com/speech/service/10011
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# 一般来说按次收费的更便宜，但是DoubaoStreamASR使用了大模型技术，效果更好
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type: doubao_stream
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appid: 你的火山引擎语音合成服务appid
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@@ -63,6 +63,8 @@ async def checkWakeupWords(conn, text):
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"""检查是否是唤醒词"""
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_, filtered_text = remove_punctuation_and_length(text)
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if filtered_text in conn.config.get("wakeup_words"):
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# 设置刚刚被唤醒的标志
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conn.just_woken_up = True
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await send_stt_message(conn, text)
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file = getWakeupWordFile(WAKEUP_CONFIG["file_name"])
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@@ -3,6 +3,7 @@ from core.handle.intentHandler import handle_user_intent
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from core.utils.output_counter import check_device_output_limit
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from core.handle.abortHandle import handleAbortMessage
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import time
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import asyncio
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from core.handle.sendAudioHandle import SentenceType
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from core.utils.util import audio_to_data
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@@ -12,6 +13,13 @@ TAG = __name__
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async def handleAudioMessage(conn, audio):
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# 当前片段是否有人说话
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have_voice = conn.vad.is_vad(conn, audio)
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# 如果设备刚刚被唤醒，短暂忽略VAD检测
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if hasattr(conn, "just_woken_up") and conn.just_woken_up:
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have_voice = False
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# 设置一个短暂延迟后恢复VAD检测
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asyncio.create_task(resume_vad_detection(conn))
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if have_voice:
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if conn.client_is_speaking:
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await handleAbortMessage(conn)
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@@ -21,6 +29,12 @@ async def handleAudioMessage(conn, audio):
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await conn.asr.receive_audio(conn, audio, have_voice)
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async def resume_vad_detection(conn):
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# 等待1秒后恢复VAD检测
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await asyncio.sleep(1)
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conn.just_woken_up = False
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async def startToChat(conn, text):
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if conn.need_bind:
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await check_bind_device(conn)
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@@ -61,6 +61,7 @@ async def handleTextMessage(conn, message):
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await send_tts_message(conn, "stop", None)
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conn.client_is_speaking = False
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elif is_wakeup_words:
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conn.just_woken_up = True
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# 上报纯文字数据（复用ASR上报功能，但不提供音频数据）
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enqueue_asr_report(conn, "嘿，你好呀", [])
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await startToChat(conn, "嘿，你好呀")
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@@ -78,7 +79,9 @@ async def handleTextMessage(conn, message):
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elif msg_json["type"] == "mcp":
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conn.logger.bind(tag=TAG).info(f"收到mcp消息：{message}")
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if "payload" in msg_json:
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asyncio.create_task(handle_mcp_message(conn, conn.mcp_client, msg_json["payload"]))
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asyncio.create_task(
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handle_mcp_message(conn, conn.mcp_client, msg_json["payload"])
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)
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elif msg_json["type"] == "server":
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# 记录日志时过滤敏感信息
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conn.logger.bind(tag=TAG).info(
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@@ -84,9 +84,13 @@ class ASRProviderBase(ABC):
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text_len, _ = remove_punctuation_and_length(raw_text)
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if text_len > 0:
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# 使用自定义模块进行上报
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self.stop_ws_connection()
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await startToChat(conn, raw_text)
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enqueue_asr_report(conn, raw_text, asr_audio_task)
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def stop_ws_connection(self):
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pass
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def save_audio_to_file(self, pcm_data: List[bytes], session_id: str) -> str:
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"""PCM数据保存为WAV文件"""
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module_name = __name__.split(".")[-1]
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@@ -7,22 +7,10 @@ import opuslib_next
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from core.providers.asr.base import ASRProviderBase
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from config.logger import setup_logging
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from core.providers.asr.dto.dto import InterfaceType
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import threading
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TAG = __name__
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logger = setup_logging()
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CLIENT_FULL_REQUEST = 0b0001
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CLIENT_AUDIO_ONLY_REQUEST = 0b0010
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SERVER_FULL_RESPONSE = 0b1001
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SERVER_ACK = 0b1011
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SERVER_ERROR_RESPONSE = 0b1111
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NO_SEQUENCE = 0b0000
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NEG_SEQUENCE = 0b0010
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JSON_SERIALIZATION = 0b0001
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GZIP_COMPRESSION = 0b0001
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PROTOCOL_VERSION = 0b0001
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class ASRProvider(ASRProviderBase):
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def __init__(self, config, delete_audio_file):
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@@ -31,39 +19,29 @@ class ASRProvider(ASRProviderBase):
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self.config = config
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self.text = ""
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self.max_retries = 3
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self.retry_delay = 2 # 重试延迟秒数
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self.recv_lock = asyncio.Lock() # 添加接收锁
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self.reconnect_lock = asyncio.Lock() # 添加重连锁
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self.last_reconnect_time = 0 # 上次重连时间
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self.reconnect_cooldown = 1 # 增加重连冷却时间到10秒
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self.reconnect_count = 0 # 当前重连次数
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self.max_reconnect_count = 3 # 减少最大重连次数到3次
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self.asr_thread = None # ASR监听线程
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self.thread_lock = threading.Lock() # 线程管理锁
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self.is_reconnecting = False # 添加重连状态标志
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# 添加会话管理相关属性
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self._session_lock = asyncio.Lock() # 会话操作的并发锁
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self._current_session_id = None # 当前会话ID
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self._session_started = False # 会话是否已开始
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self._session_finished = False # 会话是否已结束
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self._session_close_event = asyncio.Event() # 添加会话关闭事件
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self.retry_delay = 2
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self.decoder = opuslib_next.Decoder(16000, 1)
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self.asr_ws = None
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self.forward_task = None
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self.is_processing = False # 添加处理状态标志
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# 配置参数
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self.appid = str(config.get("appid"))
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self.cluster = config.get("cluster")
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self.access_token = config.get("access_token")
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self.boosting_table_name = config.get("boosting_table_name", "")
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self.correct_table_name = config.get("correct_table_name", "")
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self.output_dir = config.get("output_dir", "temp/")
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self.output_dir = config.get("output_dir", "tmp/")
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self.delete_audio_file = delete_audio_file
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self.ws_url = "wss://openspeech.bytedance.com/api/v2/asr"
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# 火山引擎ASR配置
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self.ws_url = "wss://openspeech.bytedance.com/api/v3/sauc/bigmodel"
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self.uid = config.get("uid", "streaming_asr_service")
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self.workflow = config.get(
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"workflow", "audio_in,resample,partition,vad,fe,decode,itn,nlu_punctuate"
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)
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self.result_type = config.get("result_type", "single")
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self.format = config.get("format", "raw")
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self.format = config.get("format", "pcm")
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self.codec = config.get("codec", "pcm")
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self.rate = config.get("sample_rate", 16000)
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self.language = config.get("language", "zh-CN")
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@@ -71,144 +49,163 @@ class ASRProvider(ASRProviderBase):
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self.channel = config.get("channel", 1)
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self.auth_method = config.get("auth_method", "token")
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self.secret = config.get("secret", "access_secret")
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self.decoder = opuslib_next.Decoder(16000, 1)
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self.asr_ws = None
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self.forward_task = None
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self.conn = None
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###################################################################################
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# 豆包流式ASR重写父类的方法--开始
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###################################################################################
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async def open_audio_channels(self, conn):
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await super().open_audio_channels(conn)
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async with self._session_lock:
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# 如果正在重连，等待重连完成
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if self.is_reconnecting:
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logger.bind(tag=TAG).info("等待当前重连完成...")
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await self._session_close_event.wait()
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self._session_close_event.clear()
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# 如果已有会话未结束，先关闭它
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if self._session_started and not self._session_finished:
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logger.bind(tag=TAG).warning(
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f"发现未关闭的会话 {self._current_session_id}，正在关闭..."
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)
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if self.asr_ws is not None:
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try:
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await self.asr_ws.close()
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except Exception as e:
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logger.bind(tag=TAG).warning(f"关闭旧连接时发生错误: {e}")
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finally:
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self.asr_ws = None
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self._session_finished = True
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self._session_close_event.set()
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# 重置会话状态
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self._current_session_id = str(uuid.uuid4())
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self._session_started = True
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self._session_finished = False
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self.is_reconnecting = True
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try:
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retry_count = 0
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while retry_count < self.max_retries:
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try:
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headers = (
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self.token_auth() if self.auth_method == "token" else None
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)
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self.asr_ws = await websockets.connect(
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self.ws_url,
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additional_headers=headers,
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max_size=1000000000,
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ping_interval=None,
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ping_timeout=None,
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close_timeout=10,
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)
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# 发送初始化请求
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request_params = self.construct_request(
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self._current_session_id
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)
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try:
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payload_bytes = str.encode(json.dumps(request_params))
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payload_bytes = gzip.compress(payload_bytes)
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full_client_request = self.generate_header()
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full_client_request.extend(
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(len(payload_bytes)).to_bytes(4, "big")
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)
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full_client_request.extend(payload_bytes)
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await self.asr_ws.send(full_client_request)
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except Exception as e:
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logger.bind(tag=TAG).error(f"发送初始化请求失败: {e}")
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raise e
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# 等待初始化响应
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try:
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init_res = await self.asr_ws.recv()
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self.parse_response(init_res)
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except Exception as e:
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logger.bind(tag=TAG).error(f"ASR服务初始化失败: {e}")
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raise e
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# 启动接收ASR结果的异步任务
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with self.thread_lock:
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if (
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self.asr_thread is None
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or not self.asr_thread.is_alive()
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):
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logger.bind(tag=TAG).info("创建新的ASR监听线程...")
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self.asr_thread = threading.Thread(
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target=self._start_monitor_asr_response_thread,
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daemon=True,
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)
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self.asr_thread.start()
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# 等待一小段时间确保线程启动
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await asyncio.sleep(0.1)
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if not self.asr_thread.is_alive():
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logger.bind(tag=TAG).error("ASR监听线程启动失败")
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raise Exception("ASR监听线程启动失败")
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logger.bind(tag=TAG).info("ASR监听线程已启动")
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return
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except websockets.exceptions.WebSocketException as e:
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retry_count += 1
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if retry_count < self.max_retries:
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logger.bind(tag=TAG).warning(
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f"WebSocket连接失败，正在进行第{retry_count}次重试: {e}"
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)
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await asyncio.sleep(self.retry_delay)
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else:
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logger.bind(tag=TAG).warning(
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f"WebSocket连接失败，已达到最大重试次数: {e}"
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)
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raise
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except Exception as e:
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logger.bind(tag=TAG).error(f"WebSocket连接发生未知错误: {e}")
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raise
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finally:
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self.is_reconnecting = False
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self._session_close_event.set()
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async def receive_audio(self, conn, audio, _):
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async def receive_audio(self, conn, audio, audio_have_voice):
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if not isinstance(audio, bytes):
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return
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try:
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# 解码opus得到PCM数据
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pcm_frame = self.decoder.decode(audio, 960)
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payload = gzip.compress(pcm_frame)
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audio_request = bytearray(self.generate_audio_default_header())
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audio_request.extend(len(payload).to_bytes(4, "big"))
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audio_request.extend(payload)
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if self.asr_ws:
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await self.asr_ws.send(audio_request)
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except Exception as e:
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logger.bind(tag=TAG).debug(f"发送音频数据时发生错误: {e}")
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conn.asr_audio.append(audio)
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if audio_have_voice == False and conn.client_have_voice == False:
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conn.asr_audio = conn.asr_audio[-10:]
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###################################################################################
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# 豆包流式ASR重写父类的方法--结束
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###################################################################################
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# 如果本次有声音，且之前没有建立连接
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if audio_have_voice and self.asr_ws is None and not self.is_processing:
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try:
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self.is_processing = True
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# 建立新的WebSocket连接
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headers = self.token_auth() if self.auth_method == "token" else None
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logger.bind(tag=TAG).info(f"正在连接ASR服务，headers: {headers}")
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self.asr_ws = await websockets.connect(
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self.ws_url,
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additional_headers=headers,
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max_size=1000000000,
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ping_interval=None,
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ping_timeout=None,
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close_timeout=10,
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)
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# 发送初始化请求
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request_params = self.construct_request(str(uuid.uuid4()))
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try:
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payload_bytes = str.encode(json.dumps(request_params))
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payload_bytes = gzip.compress(payload_bytes)
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full_client_request = self.generate_header()
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full_client_request.extend((len(payload_bytes)).to_bytes(4, "big"))
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full_client_request.extend(payload_bytes)
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logger.bind(tag=TAG).info(f"发送初始化请求: {request_params}")
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await self.asr_ws.send(full_client_request)
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# 等待初始化响应
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init_res = await self.asr_ws.recv()
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result = self.parse_response(init_res)
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logger.bind(tag=TAG).info(f"收到初始化响应: {result}")
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# 检查初始化响应
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if "code" in result and result["code"] != 1000:
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error_msg = f"ASR服务初始化失败: {result.get('payload_msg', {}).get('message', '未知错误')}"
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if "payload_msg" in result:
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error_msg += f"\n详细错误信息: {json.dumps(result['payload_msg'], ensure_ascii=False)}"
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logger.bind(tag=TAG).error(error_msg)
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raise Exception(error_msg)
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except Exception as e:
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logger.bind(tag=TAG).error(f"发送初始化请求失败: {str(e)}")
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if hasattr(e, "__cause__") and e.__cause__:
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logger.bind(tag=TAG).error(f"错误原因: {str(e.__cause__)}")
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raise e
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# 启动接收ASR结果的异步任务
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self.forward_task = asyncio.create_task(self._forward_asr_results(conn))
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# 发送缓存的音频数据
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if conn.asr_audio and len(conn.asr_audio) > 0:
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logger.bind(tag=TAG).info(
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f"发送缓存音频数据: {len(conn.asr_audio)}"
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)
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for cached_audio in conn.asr_audio[-10:]:
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try:
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pcm_frame = self.decoder.decode(cached_audio, 960)
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payload = gzip.compress(pcm_frame)
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audio_request = bytearray(
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self.generate_audio_default_header()
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)
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audio_request.extend(len(payload).to_bytes(4, "big"))
|
||||
audio_request.extend(payload)
|
||||
await self.asr_ws.send(audio_request)
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).info(
|
||||
f"发送缓存音频数据时发生错误: {e}"
|
||||
)
|
||||
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).error(f"建立ASR连接失败: {str(e)}")
|
||||
if hasattr(e, "__cause__") and e.__cause__:
|
||||
logger.bind(tag=TAG).error(f"错误原因: {str(e.__cause__)}")
|
||||
if self.asr_ws:
|
||||
await self.asr_ws.close()
|
||||
self.asr_ws = None
|
||||
self.is_processing = False
|
||||
return
|
||||
|
||||
# 发送当前音频数据
|
||||
if self.asr_ws and self.is_processing:
|
||||
try:
|
||||
pcm_frame = self.decoder.decode(audio, 960)
|
||||
payload = gzip.compress(pcm_frame)
|
||||
audio_request = bytearray(self.generate_audio_default_header())
|
||||
audio_request.extend(len(payload).to_bytes(4, "big"))
|
||||
audio_request.extend(payload)
|
||||
await self.asr_ws.send(audio_request)
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).info(f"发送音频数据时发生错误: {e}")
|
||||
|
||||
async def _forward_asr_results(self, conn):
|
||||
try:
|
||||
while self.asr_ws and not conn.stop_event.is_set():
|
||||
try:
|
||||
response = await self.asr_ws.recv()
|
||||
result = self.parse_response(response)
|
||||
logger.bind(tag=TAG).debug(f"收到ASR结果: {result}")
|
||||
|
||||
if "payload_msg" in result:
|
||||
payload = result["payload_msg"]
|
||||
if "result" in payload:
|
||||
utterances = payload["result"].get("utterances", [])
|
||||
for utterance in utterances:
|
||||
if utterance.get("definite", False):
|
||||
self.text = utterance["text"]
|
||||
logger.bind(tag=TAG).info(
|
||||
f"识别到文本: {self.text}"
|
||||
)
|
||||
conn.reset_vad_states()
|
||||
await self.handle_voice_stop(conn, None)
|
||||
break
|
||||
elif "error" in payload:
|
||||
error_msg = payload.get("error", "未知错误")
|
||||
logger.bind(tag=TAG).error(f"ASR服务返回错误: {error_msg}")
|
||||
break
|
||||
|
||||
except websockets.ConnectionClosed:
|
||||
logger.bind(tag=TAG).info("ASR服务连接已关闭")
|
||||
self.is_processing = False
|
||||
break
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).error(f"处理ASR结果时发生错误: {str(e)}")
|
||||
if hasattr(e, "__cause__") and e.__cause__:
|
||||
logger.bind(tag=TAG).error(f"错误原因: {str(e.__cause__)}")
|
||||
self.is_processing = False
|
||||
break
|
||||
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).error(f"ASR结果转发任务发生错误: {str(e)}")
|
||||
if hasattr(e, "__cause__") and e.__cause__:
|
||||
logger.bind(tag=TAG).error(f"错误原因: {str(e.__cause__)}")
|
||||
finally:
|
||||
if self.asr_ws:
|
||||
await self.asr_ws.close()
|
||||
self.asr_ws = None
|
||||
self.is_processing = False
|
||||
|
||||
def stop_ws_connection(self):
|
||||
if self.asr_ws:
|
||||
asyncio.create_task(self.asr_ws.close())
|
||||
self.asr_ws = None
|
||||
self.is_processing = False
|
||||
|
||||
def construct_request(self, reqid):
|
||||
req = {
|
||||
@@ -226,6 +223,7 @@ class ASRProvider(ASRProviderBase):
|
||||
"sequence": 1,
|
||||
"boosting_table_name": self.boosting_table_name,
|
||||
"correct_table_name": self.correct_table_name,
|
||||
"end_window_size": 200,
|
||||
},
|
||||
"audio": {
|
||||
"format": self.format,
|
||||
@@ -234,31 +232,33 @@ class ASRProvider(ASRProviderBase):
|
||||
"language": self.language,
|
||||
"bits": self.bits,
|
||||
"channel": self.channel,
|
||||
"sample_rate": self.rate,
|
||||
},
|
||||
}
|
||||
logger.bind(tag=TAG).info(
|
||||
f"构造请求参数: {json.dumps(req, ensure_ascii=False)}"
|
||||
)
|
||||
return req
|
||||
|
||||
def token_auth(self):
|
||||
return {"Authorization": f"Bearer; {self.access_token}"}
|
||||
return {
|
||||
"X-Api-App-Key": self.appid,
|
||||
"X-Api-Access-Key": self.access_token,
|
||||
"X-Api-Resource-Id": "volc.bigasr.sauc.duration",
|
||||
"X-Api-Connect-Id": str(uuid.uuid4()),
|
||||
"Host": "openspeech.bytedance.com",
|
||||
}
|
||||
|
||||
def generate_header(
|
||||
self,
|
||||
version=PROTOCOL_VERSION,
|
||||
message_type=CLIENT_FULL_REQUEST,
|
||||
message_type_specific_flags=NO_SEQUENCE,
|
||||
serial_method=JSON_SERIALIZATION,
|
||||
compression_type=GZIP_COMPRESSION,
|
||||
version=0x01,
|
||||
message_type=0x01,
|
||||
message_type_specific_flags=0x00,
|
||||
serial_method=0x01,
|
||||
compression_type=0x01,
|
||||
reserved_data=0x00,
|
||||
extension_header: bytes = b"",
|
||||
):
|
||||
"""
|
||||
生成协议头：
|
||||
- 第1字节：高4位：协议版本，低4位：头部大小（单位 4 字节）
|
||||
- 第2字节：高4位：消息类型，低4位：消息类型特定标志
|
||||
- 第3字节：高4位：序列化方式，低4位：压缩方式
|
||||
- 第4字节：保留字段
|
||||
- 后续：扩展头（如果有）
|
||||
"""
|
||||
header = bytearray()
|
||||
header_size = int(len(extension_header) / 4) + 1
|
||||
header.append((version << 4) | header_size)
|
||||
@@ -268,267 +268,72 @@ class ASRProvider(ASRProviderBase):
|
||||
header.extend(extension_header)
|
||||
return header
|
||||
|
||||
def generate_full_default_header(self):
|
||||
# full client request 默认头
|
||||
return self.generate_header(
|
||||
version=PROTOCOL_VERSION,
|
||||
message_type=CLIENT_FULL_REQUEST,
|
||||
message_type_specific_flags=NO_SEQUENCE,
|
||||
serial_method=JSON_SERIALIZATION,
|
||||
compression_type=GZIP_COMPRESSION,
|
||||
)
|
||||
|
||||
def generate_audio_default_header(self):
|
||||
# 普通音频片段请求
|
||||
return self.generate_header(
|
||||
version=PROTOCOL_VERSION,
|
||||
message_type=CLIENT_AUDIO_ONLY_REQUEST,
|
||||
message_type_specific_flags=NO_SEQUENCE,
|
||||
serial_method=JSON_SERIALIZATION,
|
||||
compression_type=GZIP_COMPRESSION,
|
||||
version=0x01,
|
||||
message_type=0x02,
|
||||
message_type_specific_flags=0x00,
|
||||
serial_method=0x01,
|
||||
compression_type=0x01,
|
||||
)
|
||||
|
||||
def generate_last_audio_default_header(self):
|
||||
# 最后一个音频片段标志
|
||||
return self.generate_header(
|
||||
version=PROTOCOL_VERSION,
|
||||
message_type=CLIENT_AUDIO_ONLY_REQUEST,
|
||||
message_type_specific_flags=NEG_SEQUENCE, # 用 NEG_SEQUENCE 表示结束
|
||||
serial_method=JSON_SERIALIZATION,
|
||||
compression_type=GZIP_COMPRESSION,
|
||||
version=0x01,
|
||||
message_type=0x02,
|
||||
message_type_specific_flags=0x02,
|
||||
serial_method=0x01,
|
||||
compression_type=0x01,
|
||||
)
|
||||
|
||||
def _start_monitor_asr_response_thread(self):
|
||||
# 初始化链接
|
||||
def parse_response(self, res: bytes) -> dict:
|
||||
try:
|
||||
with self.thread_lock:
|
||||
if self.conn is None or self.conn.loop is None:
|
||||
logger.bind(tag=TAG).error(
|
||||
"无法启动ASR监听线程：conn或loop未初始化"
|
||||
)
|
||||
return
|
||||
# 检查响应长度
|
||||
if len(res) < 4:
|
||||
logger.bind(tag=TAG).error(f"响应数据长度不足: {len(res)}")
|
||||
return {"error": "响应数据长度不足"}
|
||||
|
||||
try:
|
||||
logger.bind(tag=TAG).info("开始启动ASR监听...")
|
||||
asyncio.run_coroutine_threadsafe(
|
||||
self._forward_asr_results(), loop=self.conn.loop
|
||||
)
|
||||
logger.bind(tag=TAG).info("ASR监听已启动")
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).error(f"启动ASR监听线程失败: {e}")
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).error(f"ASR监听线程发生未预期的错误: {e}")
|
||||
# 获取消息头
|
||||
header = res[:4]
|
||||
message_type = header[1] >> 4
|
||||
|
||||
async def _forward_asr_results(self):
|
||||
try:
|
||||
while not self.conn.stop_event.is_set():
|
||||
try:
|
||||
if self.asr_ws is None:
|
||||
# 检查是否需要重连
|
||||
async with self.reconnect_lock:
|
||||
current_time = asyncio.get_event_loop().time()
|
||||
if (
|
||||
current_time - self.last_reconnect_time
|
||||
< self.reconnect_cooldown
|
||||
):
|
||||
await asyncio.sleep(1)
|
||||
continue
|
||||
# 如果是错误响应
|
||||
if message_type == 0x0F: # SERVER_ERROR_RESPONSE
|
||||
code = int.from_bytes(header[4:8], "big", signed=False)
|
||||
error_msg = res[8:].decode("utf-8")
|
||||
return {"code": code, "error": error_msg}
|
||||
|
||||
if self.reconnect_count >= self.max_reconnect_count:
|
||||
logger.bind(tag=TAG).error(
|
||||
"达到最大重连次数限制，停止重连"
|
||||
)
|
||||
await asyncio.sleep(self.reconnect_cooldown)
|
||||
self.reconnect_count = 0
|
||||
continue
|
||||
|
||||
self.last_reconnect_time = current_time
|
||||
self.reconnect_count += 1
|
||||
logger.bind(tag=TAG).info(
|
||||
f"尝试重新连接ASR服务... (第{self.reconnect_count}次)"
|
||||
)
|
||||
await self.open_audio_channels(self.conn)
|
||||
continue
|
||||
|
||||
# 使用锁来确保同一时间只有一个协程在接收数据
|
||||
async with self.recv_lock:
|
||||
response = await self.asr_ws.recv()
|
||||
result = self.parse_response(response)
|
||||
|
||||
# 检查是否需要重连
|
||||
if result.get("need_reconnect", False):
|
||||
logger.bind(tag=TAG).info(
|
||||
"检测到需要重连的错误，准备重新连接..."
|
||||
)
|
||||
if self.asr_ws is not None:
|
||||
try:
|
||||
await self.asr_ws.close()
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).warning(
|
||||
f"关闭旧连接时发生错误: {e}"
|
||||
)
|
||||
finally:
|
||||
self.asr_ws = None
|
||||
continue
|
||||
|
||||
if "payload_msg" in result:
|
||||
if "result" in result["payload_msg"]:
|
||||
# 检查是否有utterances并且definite为True
|
||||
utterances = result["payload_msg"]["result"][0].get(
|
||||
"utterances", []
|
||||
)
|
||||
for utterance in utterances:
|
||||
if utterance.get("definite", False):
|
||||
self.text = utterance["text"]
|
||||
await self.handle_voice_stop(None)
|
||||
break
|
||||
|
||||
except websockets.ConnectionClosed:
|
||||
logger.bind(tag=TAG).debug("ASR服务连接已关闭，准备重连...")
|
||||
# 确保关闭旧连接
|
||||
if self.asr_ws is not None:
|
||||
try:
|
||||
await self.asr_ws.close()
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).warning(f"关闭旧连接时发生错误: {e}")
|
||||
finally:
|
||||
self.asr_ws = None
|
||||
|
||||
# 等待冷却时间
|
||||
await asyncio.sleep(self.reconnect_cooldown)
|
||||
continue
|
||||
|
||||
except Exception as e:
|
||||
if not self.conn.stop_event.is_set():
|
||||
logger.bind(tag=TAG).error(f"ASR监听发生错误: {e}")
|
||||
await asyncio.sleep(self.retry_delay)
|
||||
continue
|
||||
# 获取JSON数据（跳过12字节头部）
|
||||
try:
|
||||
json_data = res[12:].decode("utf-8")
|
||||
result = json.loads(json_data)
|
||||
logger.bind(tag=TAG).debug(f"成功解析JSON响应: {result}")
|
||||
return {"payload_msg": result}
|
||||
except (UnicodeDecodeError, json.JSONDecodeError) as e:
|
||||
logger.bind(tag=TAG).error(f"JSON解析失败: {str(e)}")
|
||||
logger.bind(tag=TAG).error(f"原始数据: {res}")
|
||||
raise
|
||||
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).error(f"ASR监听线程发生错误: {e}")
|
||||
# 确保在发生严重错误时也能继续尝试重连
|
||||
if not self.conn.stop_event.is_set():
|
||||
await asyncio.sleep(self.retry_delay)
|
||||
await self._forward_asr_results() # 递归重试
|
||||
logger.bind(tag=TAG).error(f"解析响应失败: {str(e)}")
|
||||
logger.bind(tag=TAG).error(f"原始响应数据: {res.hex()}")
|
||||
raise
|
||||
|
||||
async def speech_to_text(self, opus_data, session_id, audio_format):
|
||||
result = self.text
|
||||
self.text = "" # 清空text
|
||||
return result, None
|
||||
|
||||
def parse_response(self, res: bytes) -> dict:
|
||||
"""
|
||||
解析 ASR 服务返回的二进制响应。
|
||||
根据协议格式解析头部和 payload，若采用 GZIP 压缩则先解压，再根据 JSON 反序列化。
|
||||
"""
|
||||
protocol_version = res[0] >> 4
|
||||
header_size = res[0] & 0x0F
|
||||
message_type = res[1] >> 4
|
||||
serialization_method = res[2] >> 4
|
||||
message_compression = res[2] & 0x0F
|
||||
payload = res[header_size * 4 :]
|
||||
result = {}
|
||||
payload_msg = None
|
||||
payload_size = 0
|
||||
|
||||
if message_type == SERVER_FULL_RESPONSE:
|
||||
payload_size = int.from_bytes(payload[:4], "big", signed=True)
|
||||
payload_msg = payload[4:]
|
||||
elif message_type == SERVER_ACK:
|
||||
seq = int.from_bytes(payload[:4], "big", signed=True)
|
||||
result["seq"] = seq
|
||||
if len(payload) >= 8:
|
||||
payload_size = int.from_bytes(payload[4:8], "big", signed=False)
|
||||
payload_msg = payload[8:]
|
||||
elif message_type == SERVER_ERROR_RESPONSE:
|
||||
code = int.from_bytes(payload[:4], "big", signed=False)
|
||||
result["code"] = code
|
||||
payload_size = int.from_bytes(payload[4:8], "big", signed=False)
|
||||
payload_msg = payload[8:]
|
||||
|
||||
if payload_msg is None:
|
||||
return result
|
||||
if message_compression == GZIP_COMPRESSION:
|
||||
payload_msg = gzip.decompress(payload_msg)
|
||||
if serialization_method == JSON_SERIALIZATION:
|
||||
payload_msg = json.loads(payload_msg.decode("utf-8"))
|
||||
else:
|
||||
payload_msg = payload_msg.decode("utf-8")
|
||||
result["payload_msg"] = payload_msg
|
||||
result["payload_size"] = payload_size
|
||||
|
||||
# 错误码处理
|
||||
if "code" in result:
|
||||
error_code = result["code"]
|
||||
error_message = ""
|
||||
|
||||
if error_code == 1000:
|
||||
error_message = "成功"
|
||||
elif error_code == 1001:
|
||||
error_message = "请求参数无效：请求参数缺失必需字段/字段值无效/重复请求"
|
||||
elif error_code == 1002:
|
||||
error_message = "无访问权限：token无效/过期/无权访问指定服务"
|
||||
elif error_code == 1003:
|
||||
error_message = "访问超频：当前appid访问QPS超出设定阈值"
|
||||
elif error_code == 1004:
|
||||
error_message = "访问超额：当前appid访问次数超出限制"
|
||||
elif error_code == 1005:
|
||||
error_message = "服务器繁忙：服务过载，无法处理当前请求"
|
||||
elif error_code == 1010:
|
||||
error_message = "音频过长：音频数据时长超出阈值"
|
||||
elif error_code == 1011:
|
||||
error_message = "音频过大：音频数据大小超出阈值"
|
||||
elif error_code == 1012:
|
||||
error_message = "音频格式无效：音频header有误/无法进行音频解码"
|
||||
elif error_code == 1013:
|
||||
error_message = "音频静音：音频未识别出任何文本结果"
|
||||
elif error_code >= 1020 and error_code <= 1022:
|
||||
error_message = "识别相关错误：需要重连"
|
||||
if error_code == 1020:
|
||||
error_message = "识别等待超时：等待下一包就绪超时"
|
||||
elif error_code == 1021:
|
||||
error_message = "识别处理超时：识别处理过程超时"
|
||||
elif error_code == 1022:
|
||||
error_message = "识别错误：识别过程中发生错误"
|
||||
else:
|
||||
error_message = "未知错误：未归类错误"
|
||||
|
||||
logger.bind(tag=TAG).debug(
|
||||
f"ASR错误: {error_message} (错误码: {error_code})"
|
||||
)
|
||||
|
||||
# 如果是识别相关错误，标记需要重连
|
||||
if error_code >= 1020 or error_code == 1001:
|
||||
result["need_reconnect"] = True
|
||||
|
||||
return result
|
||||
|
||||
async def close_session(self):
|
||||
"""关闭当前会话"""
|
||||
async with self._session_lock:
|
||||
if not self._session_started:
|
||||
logger.bind(tag=TAG).warning("尝试关闭未开始的会话")
|
||||
return
|
||||
|
||||
if self._session_finished:
|
||||
logger.bind(tag=TAG).warning(
|
||||
f"会话 {self._current_session_id} 已经关闭"
|
||||
)
|
||||
return
|
||||
|
||||
try:
|
||||
if self.asr_ws is not None:
|
||||
await self.asr_ws.close()
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).warning(f"关闭WebSocket连接时发生错误: {e}")
|
||||
finally:
|
||||
self.asr_ws = None
|
||||
self._session_finished = True
|
||||
self._session_started = False
|
||||
self._current_session_id = None
|
||||
# 重置重连计数
|
||||
self.reconnect_count = 0
|
||||
|
||||
async def close(self):
|
||||
"""资源清理方法"""
|
||||
await self.close_session()
|
||||
if self.asr_ws:
|
||||
await self.asr_ws.close()
|
||||
self.asr_ws = None
|
||||
if self.forward_task:
|
||||
self.forward_task.cancel()
|
||||
try:
|
||||
await self.forward_task
|
||||
except asyncio.CancelledError:
|
||||
pass
|
||||
self.forward_task = None
|
||||
self.is_processing = False
|
||||
|
||||
Reference in New Issue
Block a user