mirror of
https://github.com/xinnan-tech/xiaozhi-esp32-server.git
synced 2026-07-30 17:03:56 +08:00
update:兼容豆包流式ASR
This commit is contained in:
hrz
@@ -1,265 +1,431 @@
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import time
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import os
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import uuid
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import json
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import gzip
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import uuid
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import asyncio
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import websockets
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from config.logger import setup_logging
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from typing import Optional, Tuple, List
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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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NO_SEQUENCE = 0b0000
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NEG_SEQUENCE = 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_SERIALIZATION = 0b0000
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JSON = 0b0001
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THRIFT = 0b0011
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CUSTOM_TYPE = 0b1111
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NO_COMPRESSION = 0b0000
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GZIP = 0b0001
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CUSTOM_COMPRESSION = 0b1111
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def parse_response(res):
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"""
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protocol_version(4 bits), header_size(4 bits),
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message_type(4 bits), message_type_specific_flags(4 bits)
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serialization_method(4 bits) message_compression(4 bits)
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reserved （8bits) 保留字段
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header_extensions 扩展头(大小等于 8 * 4 * (header_size - 1) )
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payload 类似与http 请求体
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"""
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protocol_version = res[0] >> 4
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header_size = res[0] & 0x0F
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message_type = res[1] >> 4
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message_type_specific_flags = res[1] & 0x0F
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serialization_method = res[2] >> 4
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message_compression = res[2] & 0x0F
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reserved = res[3]
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header_extensions = res[4 : header_size * 4]
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payload = res[header_size * 4 :]
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result = {}
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payload_msg = None
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payload_size = 0
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if message_type == SERVER_FULL_RESPONSE:
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payload_size = int.from_bytes(payload[:4], "big", signed=True)
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payload_msg = payload[4:]
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elif message_type == SERVER_ACK:
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seq = int.from_bytes(payload[:4], "big", signed=True)
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result["seq"] = seq
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if len(payload) >= 8:
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payload_size = int.from_bytes(payload[4:8], "big", signed=False)
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payload_msg = payload[8:]
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elif message_type == SERVER_ERROR_RESPONSE:
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code = int.from_bytes(payload[:4], "big", signed=False)
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result["code"] = code
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payload_size = int.from_bytes(payload[4:8], "big", signed=False)
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payload_msg = payload[8:]
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if payload_msg is None:
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return result
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if message_compression == GZIP:
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payload_msg = gzip.decompress(payload_msg)
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if serialization_method == JSON:
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payload_msg = json.loads(str(payload_msg, "utf-8"))
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elif serialization_method != NO_SERIALIZATION:
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payload_msg = str(payload_msg, "utf-8")
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result["payload_msg"] = payload_msg
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result["payload_size"] = payload_size
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return result
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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: dict, delete_audio_file: bool):
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def __init__(self, config, delete_audio_file):
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super().__init__()
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self.appid = config.get("appid")
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self.interface_type = InterfaceType.STREAM
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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.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")
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self.output_dir = config.get("output_dir", "temp/")
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self.delete_audio_file = delete_audio_file
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self.host = "openspeech.bytedance.com"
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self.ws_url = f"wss://{self.host}/api/v2/asr"
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self.success_code = 1000
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self.seg_duration = 15000
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self.ws_url = "wss://openspeech.bytedance.com/api/v2/asr"
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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.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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self.bits = config.get("bits", 16)
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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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self.asr_thread = None
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# 确保输出目录存在
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os.makedirs(self.output_dir, exist_ok=True)
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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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@staticmethod
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def _generate_header(
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message_type=CLIENT_FULL_REQUEST, message_type_specific_flags=NO_SEQUENCE
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) -> bytearray:
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"""Generate protocol header."""
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header = bytearray()
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header_size = 1
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header.append((0b0001 << 4) | header_size) # Protocol version
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header.append((message_type << 4) | message_type_specific_flags)
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header.append((0b0001 << 4) | 0b0001) # JSON serialization & GZIP compression
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header.append(0x00) # reserved
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return header
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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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# 确保关闭旧的连接
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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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def _construct_request(self, reqid) -> dict:
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"""Construct the request payload."""
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return {
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headers = self.token_auth() if self.auth_method == "token" else None
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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, # 禁用ping，因为服务器可能不支持
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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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await self.asr_ws.send(full_client_request)
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logger.bind(tag=TAG).debug(f"发送初始化请求: {request_params}")
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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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logger.bind(tag=TAG).debug(f"收到原始响应: {init_res}")
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result = self.parse_response(init_res)
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logger.bind(tag=TAG).info(f"ASR服务初始化响应: {result}")
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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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asr_priority = threading.Thread(
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target=self._start_monitor_asr_response_thread, daemon=True
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)
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asr_priority.start()
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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).error(
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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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async def receive_audio(self, audio, _):
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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).error(f"发送音频数据时发生错误: {e}")
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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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def construct_request(self, reqid):
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req = {
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"app": {
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"appid": f"{self.appid}",
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"appid": self.appid,
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"cluster": self.cluster,
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"token": self.access_token,
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},
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"user": {
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"uid": str(uuid.uuid4()),
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},
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"user": {"uid": self.uid},
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"request": {
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"reqid": reqid,
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"show_utterances": False,
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"workflow": self.workflow,
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"show_utterances": True,
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"result_type": self.result_type,
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"sequence": 1,
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"boosting_table_name": self.boosting_table_name,
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"correct_table_name": self.correct_table_name,
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},
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"audio": {
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"format": "raw",
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"rate": 16000,
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"language": "zh-CN",
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"bits": 16,
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"channel": 1,
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"codec": "raw",
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"format": self.format,
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"codec": self.codec,
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"rate": self.rate,
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"language": self.language,
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"bits": self.bits,
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"channel": self.channel,
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},
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}
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return req
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async def _send_request(
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self, audio_data: List[bytes], segment_size: int
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) -> Optional[str]:
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"""Send request to Volcano ASR service."""
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def token_auth(self):
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return {"Authorization": f"Bearer; {self.access_token}"}
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def generate_header(
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self,
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version=PROTOCOL_VERSION,
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message_type=CLIENT_FULL_REQUEST,
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message_type_specific_flags=NO_SEQUENCE,
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serial_method=JSON_SERIALIZATION,
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compression_type=GZIP_COMPRESSION,
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reserved_data=0x00,
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extension_header: bytes = b"",
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):
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"""
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生成协议头：
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- 第1字节：高4位：协议版本，低4位：头部大小（单位 4 字节）
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- 第2字节：高4位：消息类型，低4位：消息类型特定标志
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- 第3字节：高4位：序列化方式，低4位：压缩方式
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- 第4字节：保留字段
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- 后续：扩展头（如果有）
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"""
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header = bytearray()
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header_size = int(len(extension_header) / 4) + 1
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header.append((version << 4) | header_size)
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header.append((message_type << 4) | message_type_specific_flags)
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header.append((serial_method << 4) | compression_type)
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header.append(reserved_data)
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header.extend(extension_header)
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return header
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def generate_full_default_header(self):
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# full client request 默认头
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return self.generate_header(
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version=PROTOCOL_VERSION,
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message_type=CLIENT_FULL_REQUEST,
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message_type_specific_flags=NO_SEQUENCE,
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serial_method=JSON_SERIALIZATION,
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compression_type=GZIP_COMPRESSION,
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)
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def generate_audio_default_header(self):
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# 普通音频片段请求
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return self.generate_header(
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version=PROTOCOL_VERSION,
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message_type=CLIENT_AUDIO_ONLY_REQUEST,
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message_type_specific_flags=NO_SEQUENCE,
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serial_method=JSON_SERIALIZATION,
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compression_type=GZIP_COMPRESSION,
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)
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def generate_last_audio_default_header(self):
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# 最后一个音频片段标志
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return self.generate_header(
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version=PROTOCOL_VERSION,
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message_type=CLIENT_AUDIO_ONLY_REQUEST,
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message_type_specific_flags=NEG_SEQUENCE, # 用 NEG_SEQUENCE 表示结束
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serial_method=JSON_SERIALIZATION,
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compression_type=GZIP_COMPRESSION,
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)
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def _start_monitor_asr_response_thread(self):
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# 初始化链接
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asyncio.run_coroutine_threadsafe(
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self._forward_asr_results(), loop=self.conn.loop
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)
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async def _forward_asr_results(self):
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try:
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auth_header = {"Authorization": "Bearer; {}".format(self.access_token)}
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async with websockets.connect(
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self.ws_url, additional_headers=auth_header
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) as websocket:
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# Prepare request data
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request_params = self._construct_request(str(uuid.uuid4()))
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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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) # payload size(4 bytes)
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full_client_request.extend(payload_bytes) # payload
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while not self.conn.stop_event.is_set():
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try:
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if self.asr_ws is None:
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logger.bind(tag=TAG).info("尝试重新连接ASR服务...")
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await self.open_audio_channels(self.conn)
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continue
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# Send header and metadata
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# full_client_request
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await websocket.send(full_client_request)
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res = await websocket.recv()
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result = parse_response(res)
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if (
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"payload_msg" in result
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and result["payload_msg"]["code"] != self.success_code
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):
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logger.bind(tag=TAG).error(f"ASR error: {result}")
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return None
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for seq, (chunk, last) in enumerate(
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self.slice_data(audio_data, segment_size), 1
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):
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if last:
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audio_only_request = self._generate_header(
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message_type=CLIENT_AUDIO_ONLY_REQUEST,
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message_type_specific_flags=NEG_SEQUENCE,
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# 使用锁来确保同一时间只有一个协程在接收数据
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async with self.recv_lock:
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response = await self.asr_ws.recv()
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result = self.parse_response(response)
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# 检查是否需要重连
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if result.get("need_reconnect", False):
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logger.bind(tag=TAG).info(
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"检测到需要重连的错误，准备重新连接..."
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)
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else:
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audio_only_request = self._generate_header(
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message_type=CLIENT_AUDIO_ONLY_REQUEST
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)
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payload_bytes = gzip.compress(chunk)
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audio_only_request.extend(
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(len(payload_bytes)).to_bytes(4, "big")
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) # payload size(4 bytes)
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audio_only_request.extend(payload_bytes) # payload
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# Send audio data
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await websocket.send(audio_only_request)
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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(
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f"关闭旧连接时发生错误: {e}"
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)
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finally:
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self.asr_ws = None
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continue
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# Receive response
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||||
response = await websocket.recv()
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result = parse_response(response)
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|
||||
if (
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"payload_msg" in result
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||||
and result["payload_msg"]["code"] == self.success_code
|
||||
):
|
||||
if len(result["payload_msg"]["result"]) > 0:
|
||||
return result["payload_msg"]["result"][0]["text"]
|
||||
return None
|
||||
else:
|
||||
logger.bind(tag=TAG).error(f"ASR error: {result}")
|
||||
return None
|
||||
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()
|
||||
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
|
||||
|
||||
retry_count = 0
|
||||
while (
|
||||
retry_count < self.max_retries
|
||||
and not self.conn.stop_event.is_set()
|
||||
):
|
||||
try:
|
||||
logger.bind(tag=TAG).info(
|
||||
f"正在进行第{retry_count + 1}次重连尝试..."
|
||||
)
|
||||
await self.open_audio_channels(self.conn)
|
||||
break
|
||||
except Exception as e:
|
||||
retry_count += 1
|
||||
if retry_count < self.max_retries:
|
||||
logger.bind(tag=TAG).warning(
|
||||
f"重连失败，等待{self.retry_delay}秒后重试: {e}"
|
||||
)
|
||||
await asyncio.sleep(self.retry_delay)
|
||||
else:
|
||||
logger.bind(tag=TAG).error(
|
||||
f"重连失败，已达到最大重试次数: {e}"
|
||||
)
|
||||
await asyncio.sleep(
|
||||
self.retry_delay
|
||||
) # 继续等待，以便后续重试
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).error(f"处理ASR结果时发生错误: {e}")
|
||||
if not self.conn.stop_event.is_set():
|
||||
await asyncio.sleep(2) # 增加重试延迟
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).error(f"ASR request failed: {e}", exc_info=True)
|
||||
return None
|
||||
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
|
||||
|
||||
@staticmethod
|
||||
def slice_data(data: bytes, chunk_size: int) -> (list, bool):
|
||||
async def speech_to_text(self, opus_data, session_id):
|
||||
result = self.text
|
||||
self.text = "" # 清空text
|
||||
return result, None
|
||||
|
||||
def parse_response(self, res: bytes) -> dict:
|
||||
"""
|
||||
slice data
|
||||
:param data: wav data
|
||||
:param chunk_size: the segment size in one request
|
||||
:return: segment data, last flag
|
||||
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。
|
||||
"""
|
||||
data_len = len(data)
|
||||
offset = 0
|
||||
while offset + chunk_size < data_len:
|
||||
yield data[offset : offset + chunk_size], False
|
||||
offset += chunk_size
|
||||
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:
|
||||
yield data[offset:data_len], True
|
||||
payload_msg = payload_msg.decode("utf-8")
|
||||
result["payload_msg"] = payload_msg
|
||||
result["payload_size"] = payload_size
|
||||
|
||||
async def speech_to_text(
|
||||
self, opus_data: List[bytes], session_id: str
|
||||
) -> Tuple[Optional[str], Optional[str]]:
|
||||
"""将语音数据转换为文本"""
|
||||
# 错误码处理
|
||||
if "code" in result:
|
||||
error_code = result["code"]
|
||||
error_message = ""
|
||||
|
||||
file_path = None
|
||||
try:
|
||||
# 合并所有opus数据包
|
||||
if self.audio_format == "pcm":
|
||||
pcm_data = opus_data
|
||||
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:
|
||||
pcm_data = self.decode_opus(opus_data)
|
||||
combined_pcm_data = b"".join(pcm_data)
|
||||
error_message = "未知错误：未归类错误"
|
||||
|
||||
# 判断是否保存为WAV文件
|
||||
if self.delete_audio_file:
|
||||
pass
|
||||
else:
|
||||
file_path = self.save_audio_to_file(pcm_data, session_id)
|
||||
logger.bind(tag=TAG).debug(
|
||||
f"ASR错误: {error_message} (错误码: {error_code})"
|
||||
)
|
||||
|
||||
# 直接使用PCM数据
|
||||
# 计算分段大小 (单声道, 16bit, 16kHz采样率)
|
||||
size_per_sec = 1 * 2 * 16000 # nchannels * sampwidth * framerate
|
||||
segment_size = int(size_per_sec * self.seg_duration / 1000)
|
||||
# 如果是识别相关错误（>=1020），标记需要重连
|
||||
if error_code >= 1020:
|
||||
result["need_reconnect"] = True
|
||||
|
||||
# 语音识别
|
||||
start_time = time.time()
|
||||
text = await self._send_request(combined_pcm_data, segment_size)
|
||||
if text:
|
||||
logger.bind(tag=TAG).debug(
|
||||
f"语音识别耗时: {time.time() - start_time:.3f}s | 结果: {text}"
|
||||
)
|
||||
return text, file_path
|
||||
return "", file_path
|
||||
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).error(f"语音识别失败: {e}", exc_info=True)
|
||||
return "", file_path
|
||||
return result
|
||||
|
||||
Reference in New Issue
Block a user