update:区分豆包ASR按次收费和按时收费接口

This commit is contained in:
hrz
2025-06-03 17:30:35 +08:00
parent 3657f6ce75
commit 610fa4d101
9 changed files with 806 additions and 478 deletions
@@ -227,7 +227,7 @@ public interface Constant {
/**
* 版本号
*/
public static final String VERSION = "0.5.2";
public static final String VERSION = "0.5.4";
/**
* 无效固件URL
@@ -0,0 +1,45 @@
-- VLLM模型供应器
delete from `ai_model_provider` where id = 'SYSTEM_ASR_DoubaoStreamASR';
INSERT INTO `ai_model_provider` (`id`, `model_type`, `provider_code`, `name`, `fields`, `sort`, `creator`, `create_date`, `updater`, `update_date`) VALUES
('SYSTEM_ASR_DoubaoStreamASR', 'ASR', 'doubao_stream', '火山引擎语音识别(流式)', '[{"key":"appid","label":"应用ID","type":"string"},{"key":"access_token","label":"访问令牌","type":"string"},{"key":"cluster","label":"集群","type":"string"},{"key":"boosting_table_name","label":"热词文件名称","type":"string"},{"key":"correct_table_name","label":"替换词文件名称","type":"string"},{"key":"output_dir","label":"输出目录","type":"string"}]', 3, 1, NOW(), 1, NOW());
-- VLLM模型配置
delete from `ai_model_config` where id = 'ASR_DoubaoStreamASR';
INSERT INTO `ai_model_config` VALUES ('ASR_DoubaoStreamASR', 'ASR', 'DoubaoStreamASR', '豆包语音识别(流式)', 0, 1, '{\"type\": \"doubao_stream\", \"appid\": \"\", \"access_token\": \"\", \"cluster\": \"volcengine_input_common\", \"output_dir\": \"tmp/\"}', NULL, NULL, 3, NULL, NULL, NULL, NULL);
-- 更新豆包ASR配置说明
UPDATE `ai_model_config` SET
`doc_link` = 'https://console.volcengine.com/speech/app',
`remark` = '豆包ASR配置说明:
1. 豆包ASR和豆包(流式)ASR的区别是:豆包ASR是按次收费,豆包(流式)ASR是按时收费
2. 一般来说按次收费的更便宜,但是豆包(流式)ASR使用了大模型技术,效果更好
3. 需要在火山引擎控制台创建应用并获取appid和access_token
4. 支持中文语音识别
5. 需要网络连接
6. 输出文件保存在tmp/目录
申请步骤:
1. 访问 https://console.volcengine.com/speech/app
2. 创建新应用
3. 获取appid和access_token
4. 填入配置文件中
如需设置热词,请参考:https://www.volcengine.com/docs/6561/155738
' WHERE `id` = 'ASR_DoubaoASR';
UPDATE `ai_model_config` SET
`doc_link` = 'https://console.volcengine.com/speech/app',
`remark` = '豆包ASR配置说明:
1. 豆包ASR和豆包(流式)ASR的区别是:豆包ASR是按次收费,豆包(流式)ASR是按时收费
2. 一般来说按次收费的更便宜,但是豆包(流式)ASR使用了大模型技术,效果更好
3. 需要在火山引擎控制台创建应用并获取appid和access_token
4. 支持中文语音识别
5. 需要网络连接
6. 输出文件保存在tmp/目录
申请步骤:
1. 访问 https://console.volcengine.com/speech/app
2. 创建新应用
3. 获取appid和access_token
4. 填入配置文件中
如需设置热词,请参考:https://www.volcengine.com/docs/6561/155738
' WHERE `id` = 'ASR_DoubaoStreamASR';
+2 -2
View File
@@ -31,7 +31,7 @@
<span class="menu-text">大语言模型</span>
</el-menu-item>
<el-menu-item index="vllm">
<span class="menu-text">视觉大语言模型</span>
<span class="menu-text">视觉大模型</span>
</el-menu-item>
<el-menu-item index="intent">
<span class="menu-text">意图识别</span>
@@ -176,7 +176,7 @@ export default {
vad: '语言活动检测模型(VAD)',
asr: '语音识别模型(ASR)',
llm: '大语言模型(LLM',
vllm: '视觉大语言模型(VLLM',
vllm: '视觉大模型(VLLM',
intent: '意图识别模型(Intent)',
tts: '语音合成模型(TTS)',
memory: '记忆模型(Memory)'
+1 -1
View File
@@ -177,7 +177,7 @@ export default {
{ label: '语音活动检测(VAD)', key: 'vadModelId', type: 'VAD' },
{ label: '语音识别(ASR)', key: 'asrModelId', type: 'ASR' },
{ label: '大语言模型(LLM)', key: 'llmModelId', type: 'LLM' },
{ label: '视觉大语言模型(VLLM)', key: 'vllmModelId', type: 'VLLM' },
{ label: '视觉大模型(VLLM)', key: 'vllmModelId', type: 'VLLM' },
{ label: '意图识别(Intent)', key: 'intentModelId', type: 'Intent' },
{ label: '记忆(Memory)', key: 'memModelId', type: 'Memory' },
{ label: '语音合成(TTS)', key: 'ttsModelId', type: 'TTS' },
+15
View File
@@ -264,6 +264,8 @@ ASR:
DoubaoASR:
# 可以在这里申请相关Key等信息
# https://console.volcengine.com/speech/app
# DoubaoASR和DoubaoStreamASR的区别是:DoubaoASR是按次收费,DoubaoStreamASR是按时收费
# 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
type: doubao
appid: 你的火山引擎语音合成服务appid
access_token: 你的火山引擎语音合成服务access_token
@@ -272,6 +274,19 @@ ASR:
boosting_table_name: (选填)你的热词文件名称
correct_table_name: (选填)你的替换词文件名称
output_dir: tmp/
DoubaoStreamASR:
# 可以在这里申请相关Key等信息
# https://console.volcengine.com/speech/app
# DoubaoASR和DoubaoStreamASR的区别是:DoubaoASR是按次收费,DoubaoStreamASR是按时收费
# 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
type: doubao_stream
appid: 你的火山引擎语音合成服务appid
access_token: 你的火山引擎语音合成服务access_token
cluster: volcengine_input_common
# 热词、替换词使用流程:https://www.volcengine.com/docs/6561/155738
boosting_table_name: (选填)你的热词文件名称
correct_table_name: (选填)你的替换词文件名称
output_dir: tmp/
TencentASR:
# token申请地址:https://console.cloud.tencent.com/cam/capi
# 免费领取资源:https://console.cloud.tencent.com/asr/resourcebundle
+1 -1
View File
@@ -4,7 +4,7 @@ from loguru import logger
from config.config_loader import load_config
from config.settings import check_config_file
SERVER_VERSION = "0.5.2"
SERVER_VERSION = "0.5.4"
_logger_initialized = False
+206 -473
View File
@@ -1,534 +1,267 @@
import time
import os
import uuid
import json
import gzip
import uuid
import asyncio
import websockets
import opuslib_next
from core.providers.asr.base import ASRProviderBase
from config.logger import setup_logging
from typing import Optional, Tuple, List
from core.providers.asr.base import ASRProviderBase
from core.providers.asr.dto.dto import InterfaceType
import threading
TAG = __name__
logger = setup_logging()
CLIENT_FULL_REQUEST = 0b0001
CLIENT_AUDIO_ONLY_REQUEST = 0b0010
NO_SEQUENCE = 0b0000
NEG_SEQUENCE = 0b0010
SERVER_FULL_RESPONSE = 0b1001
SERVER_ACK = 0b1011
SERVER_ERROR_RESPONSE = 0b1111
NO_SEQUENCE = 0b0000
NEG_SEQUENCE = 0b0010
JSON_SERIALIZATION = 0b0001
GZIP_COMPRESSION = 0b0001
PROTOCOL_VERSION = 0b0001
NO_SERIALIZATION = 0b0000
JSON = 0b0001
THRIFT = 0b0011
CUSTOM_TYPE = 0b1111
NO_COMPRESSION = 0b0000
GZIP = 0b0001
CUSTOM_COMPRESSION = 0b1111
def parse_response(res):
"""
protocol_version(4 bits), header_size(4 bits),
message_type(4 bits), message_type_specific_flags(4 bits)
serialization_method(4 bits) message_compression(4 bits)
reserved 8bits) 保留字段
header_extensions 扩展头(大小等于 8 * 4 * (header_size - 1) )
payload 类似与http 请求体
"""
protocol_version = res[0] >> 4
header_size = res[0] & 0x0F
message_type = res[1] >> 4
message_type_specific_flags = res[1] & 0x0F
serialization_method = res[2] >> 4
message_compression = res[2] & 0x0F
reserved = res[3]
header_extensions = res[4 : header_size * 4]
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:
payload_msg = gzip.decompress(payload_msg)
if serialization_method == JSON:
payload_msg = json.loads(str(payload_msg, "utf-8"))
elif serialization_method != NO_SERIALIZATION:
payload_msg = str(payload_msg, "utf-8")
result["payload_msg"] = payload_msg
result["payload_size"] = payload_size
return result
class ASRProvider(ASRProviderBase):
def __init__(self, config, delete_audio_file):
def __init__(self, config: dict, delete_audio_file: bool):
super().__init__()
self.interface_type = InterfaceType.STREAM
self.config = config
self.text = ""
self.max_retries = 3
self.retry_delay = 2 # 重试延迟秒数
self.recv_lock = asyncio.Lock() # 添加接收锁
self.reconnect_lock = asyncio.Lock() # 添加重连锁
self.last_reconnect_time = 0 # 上次重连时间
self.reconnect_cooldown = 1 # 增加重连冷却时间到10秒
self.reconnect_count = 0 # 当前重连次数
self.max_reconnect_count = 3 # 减少最大重连次数到3次
self.asr_thread = None # ASR监听线程
self.thread_lock = threading.Lock() # 线程管理锁
self.is_reconnecting = False # 添加重连状态标志
# 添加会话管理相关属性
self._session_lock = asyncio.Lock() # 会话操作的并发锁
self._current_session_id = None # 当前会话ID
self._session_started = False # 会话是否已开始
self._session_finished = False # 会话是否已结束
self._session_close_event = asyncio.Event() # 添加会话关闭事件
self.appid = str(config.get("appid"))
self.interface_type = InterfaceType.NON_STREAM
self.appid = config.get("appid")
self.cluster = config.get("cluster")
self.access_token = config.get("access_token")
self.boosting_table_name = config.get("boosting_table_name", "")
self.correct_table_name = config.get("correct_table_name", "")
self.output_dir = config.get("output_dir", "temp/")
self.output_dir = config.get("output_dir")
self.delete_audio_file = delete_audio_file
self.ws_url = "wss://openspeech.bytedance.com/api/v2/asr"
self.uid = config.get("uid", "streaming_asr_service")
self.workflow = config.get(
"workflow", "audio_in,resample,partition,vad,fe,decode,itn,nlu_punctuate"
)
self.result_type = config.get("result_type", "single")
self.format = config.get("format", "raw")
self.codec = config.get("codec", "pcm")
self.rate = config.get("sample_rate", 16000)
self.language = config.get("language", "zh-CN")
self.bits = config.get("bits", 16)
self.channel = config.get("channel", 1)
self.auth_method = config.get("auth_method", "token")
self.secret = config.get("secret", "access_secret")
self.decoder = opuslib_next.Decoder(16000, 1)
self.asr_ws = None
self.forward_task = None
self.conn = None
self.host = "openspeech.bytedance.com"
self.ws_url = f"wss://{self.host}/api/v2/asr"
self.success_code = 1000
self.seg_duration = 15000
###################################################################################
# 豆包流式ASR重写父类的方法--开始
###################################################################################
async def open_audio_channels(self, conn):
await super().open_audio_channels(conn)
# 确保输出目录存在
os.makedirs(self.output_dir, exist_ok=True)
async with self._session_lock:
# 如果正在重连,等待重连完成
if self.is_reconnecting:
logger.bind(tag=TAG).info("等待当前重连完成...")
await self._session_close_event.wait()
self._session_close_event.clear()
@staticmethod
def _generate_header(
message_type=CLIENT_FULL_REQUEST, message_type_specific_flags=NO_SEQUENCE
) -> bytearray:
"""Generate protocol header."""
header = bytearray()
header_size = 1
header.append((0b0001 << 4) | header_size) # Protocol version
header.append((message_type << 4) | message_type_specific_flags)
header.append((0b0001 << 4) | 0b0001) # JSON serialization & GZIP compression
header.append(0x00) # reserved
return header
# 如果已有会话未结束,先关闭它
if self._session_started and not self._session_finished:
logger.bind(tag=TAG).warning(
f"发现未关闭的会话 {self._current_session_id},正在关闭..."
)
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
self._session_finished = True
self._session_close_event.set()
# 重置会话状态
self._current_session_id = str(uuid.uuid4())
self._session_started = True
self._session_finished = False
self.is_reconnecting = True
try:
retry_count = 0
while retry_count < self.max_retries:
try:
headers = (
self.token_auth() if self.auth_method == "token" else None
)
self.asr_ws = await websockets.connect(
self.ws_url,
additional_headers=headers,
max_size=1000000000,
ping_interval=None,
ping_timeout=None,
close_timeout=10,
)
# 发送初始化请求
request_params = self.construct_request(
self._current_session_id
)
try:
payload_bytes = str.encode(json.dumps(request_params))
payload_bytes = gzip.compress(payload_bytes)
full_client_request = self.generate_header()
full_client_request.extend(
(len(payload_bytes)).to_bytes(4, "big")
)
full_client_request.extend(payload_bytes)
await self.asr_ws.send(full_client_request)
except Exception as e:
logger.bind(tag=TAG).error(f"发送初始化请求失败: {e}")
raise e
# 等待初始化响应
try:
init_res = await self.asr_ws.recv()
self.parse_response(init_res)
except Exception as e:
logger.bind(tag=TAG).error(f"ASR服务初始化失败: {e}")
raise e
# 启动接收ASR结果的异步任务
with self.thread_lock:
if (
self.asr_thread is None
or not self.asr_thread.is_alive()
):
logger.bind(tag=TAG).info("创建新的ASR监听线程...")
self.asr_thread = threading.Thread(
target=self._start_monitor_asr_response_thread,
daemon=True,
)
self.asr_thread.start()
# 等待一小段时间确保线程启动
await asyncio.sleep(0.1)
if not self.asr_thread.is_alive():
logger.bind(tag=TAG).error("ASR监听线程启动失败")
raise Exception("ASR监听线程启动失败")
logger.bind(tag=TAG).info("ASR监听线程已启动")
return
except websockets.exceptions.WebSocketException as e:
retry_count += 1
if retry_count < self.max_retries:
logger.bind(tag=TAG).warning(
f"WebSocket连接失败,正在进行第{retry_count}次重试: {e}"
)
await asyncio.sleep(self.retry_delay)
else:
logger.bind(tag=TAG).warning(
f"WebSocket连接失败,已达到最大重试次数: {e}"
)
raise
except Exception as e:
logger.bind(tag=TAG).error(f"WebSocket连接发生未知错误: {e}")
raise
finally:
self.is_reconnecting = False
self._session_close_event.set()
async def receive_audio(self, audio, _):
if not isinstance(audio, bytes):
return
try:
# 解码opus得到PCM数据
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)
if self.asr_ws:
await self.asr_ws.send(audio_request)
except Exception as e:
logger.bind(tag=TAG).debug(f"发送音频数据时发生错误: {e}")
###################################################################################
# 豆包流式ASR重写父类的方法--结束
###################################################################################
def construct_request(self, reqid):
req = {
def _construct_request(self, reqid) -> dict:
"""Construct the request payload."""
return {
"app": {
"appid": self.appid,
"appid": f"{self.appid}",
"cluster": self.cluster,
"token": self.access_token,
},
"user": {"uid": self.uid},
"user": {
"uid": str(uuid.uuid4()),
},
"request": {
"reqid": reqid,
"workflow": self.workflow,
"show_utterances": True,
"result_type": self.result_type,
"show_utterances": False,
"sequence": 1,
"boosting_table_name": self.boosting_table_name,
"correct_table_name": self.correct_table_name,
},
"audio": {
"format": self.format,
"codec": self.codec,
"rate": self.rate,
"language": self.language,
"bits": self.bits,
"channel": self.channel,
"format": "raw",
"rate": 16000,
"language": "zh-CN",
"bits": 16,
"channel": 1,
"codec": "raw",
},
}
return req
def token_auth(self):
return {"Authorization": f"Bearer; {self.access_token}"}
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,
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)
header.append((message_type << 4) | message_type_specific_flags)
header.append((serial_method << 4) | compression_type)
header.append(reserved_data)
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,
)
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,
)
def _start_monitor_asr_response_thread(self):
# 初始化链接
async def _send_request(
self, audio_data: List[bytes], segment_size: int
) -> Optional[str]:
"""Send request to Volcano ASR service."""
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
auth_header = {"Authorization": "Bearer; {}".format(self.access_token)}
async with websockets.connect(
self.ws_url, additional_headers=auth_header
) as websocket:
# Prepare request data
request_params = self._construct_request(str(uuid.uuid4()))
payload_bytes = str.encode(json.dumps(request_params))
payload_bytes = gzip.compress(payload_bytes)
full_client_request = self._generate_header()
full_client_request.extend(
(len(payload_bytes)).to_bytes(4, "big")
) # payload size(4 bytes)
full_client_request.extend(payload_bytes) # payload
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}")
# Send header and metadata
# full_client_request
await websocket.send(full_client_request)
res = await websocket.recv()
result = parse_response(res)
if (
"payload_msg" in result
and result["payload_msg"]["code"] != self.success_code
):
logger.bind(tag=TAG).error(f"ASR error: {result}")
return None
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
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..."
for seq, (chunk, last) in enumerate(
self.slice_data(audio_data, segment_size), 1
):
if last:
audio_only_request = self._generate_header(
message_type=CLIENT_AUDIO_ONLY_REQUEST,
message_type_specific_flags=NEG_SEQUENCE,
)
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
else:
audio_only_request = self._generate_header(
message_type=CLIENT_AUDIO_ONLY_REQUEST
)
payload_bytes = gzip.compress(chunk)
audio_only_request.extend(
(len(payload_bytes)).to_bytes(4, "big")
) # payload size(4 bytes)
audio_only_request.extend(payload_bytes) # payload
# Send audio data
await websocket.send(audio_only_request)
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
# Receive response
response = await websocket.recv()
result = parse_response(response)
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
if (
"payload_msg" in result
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
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"ASR request failed: {e}", exc_info=True)
return None
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:
@staticmethod
def slice_data(data: bytes, chunk_size: int) -> (list, bool):
"""
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。
slice data
:param data: wav data
:param chunk_size: the segment size in one request
:return: segment data, last flag
"""
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"))
data_len = len(data)
offset = 0
while offset + chunk_size < data_len:
yield data[offset : offset + chunk_size], False
offset += chunk_size
else:
payload_msg = payload_msg.decode("utf-8")
result["payload_msg"] = payload_msg
result["payload_size"] = payload_size
yield data[offset:data_len], True
# 错误码处理
if "code" in result:
error_code = result["code"]
error_message = ""
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"""
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"
file_path = None
try:
# 合并所有opus数据包
if self.audio_format == "pcm":
pcm_data = opus_data
else:
error_message = "未知错误:未归类错误"
pcm_data = self.decode_opus(opus_data)
combined_pcm_data = b"".join(pcm_data)
logger.bind(tag=TAG).debug(
f"ASR错误: {error_message} (错误码: {error_code})"
)
# 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)
# 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
# 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)
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"
# 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
return text, file_path
return "", file_path
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()
except Exception as e:
logger.bind(tag=TAG).error(f"语音识别失败: {e}", exc_info=True)
return "", file_path
@@ -0,0 +1,534 @@
import json
import gzip
import uuid
import asyncio
import websockets
import opuslib_next
from core.providers.asr.base import ASRProviderBase
from config.logger import setup_logging
from core.providers.asr.dto.dto import InterfaceType
import threading
TAG = __name__
logger = setup_logging()
CLIENT_FULL_REQUEST = 0b0001
CLIENT_AUDIO_ONLY_REQUEST = 0b0010
SERVER_FULL_RESPONSE = 0b1001
SERVER_ACK = 0b1011
SERVER_ERROR_RESPONSE = 0b1111
NO_SEQUENCE = 0b0000
NEG_SEQUENCE = 0b0010
JSON_SERIALIZATION = 0b0001
GZIP_COMPRESSION = 0b0001
PROTOCOL_VERSION = 0b0001
class ASRProvider(ASRProviderBase):
def __init__(self, config, delete_audio_file):
super().__init__()
self.interface_type = InterfaceType.STREAM
self.config = config
self.text = ""
self.max_retries = 3
self.retry_delay = 2 # 重试延迟秒数
self.recv_lock = asyncio.Lock() # 添加接收锁
self.reconnect_lock = asyncio.Lock() # 添加重连锁
self.last_reconnect_time = 0 # 上次重连时间
self.reconnect_cooldown = 1 # 增加重连冷却时间到10秒
self.reconnect_count = 0 # 当前重连次数
self.max_reconnect_count = 3 # 减少最大重连次数到3次
self.asr_thread = None # ASR监听线程
self.thread_lock = threading.Lock() # 线程管理锁
self.is_reconnecting = False # 添加重连状态标志
# 添加会话管理相关属性
self._session_lock = asyncio.Lock() # 会话操作的并发锁
self._current_session_id = None # 当前会话ID
self._session_started = False # 会话是否已开始
self._session_finished = False # 会话是否已结束
self._session_close_event = asyncio.Event() # 添加会话关闭事件
self.appid = str(config.get("appid"))
self.cluster = config.get("cluster")
self.access_token = config.get("access_token")
self.boosting_table_name = config.get("boosting_table_name", "")
self.correct_table_name = config.get("correct_table_name", "")
self.output_dir = config.get("output_dir", "temp/")
self.delete_audio_file = delete_audio_file
self.ws_url = "wss://openspeech.bytedance.com/api/v2/asr"
self.uid = config.get("uid", "streaming_asr_service")
self.workflow = config.get(
"workflow", "audio_in,resample,partition,vad,fe,decode,itn,nlu_punctuate"
)
self.result_type = config.get("result_type", "single")
self.format = config.get("format", "raw")
self.codec = config.get("codec", "pcm")
self.rate = config.get("sample_rate", 16000)
self.language = config.get("language", "zh-CN")
self.bits = config.get("bits", 16)
self.channel = config.get("channel", 1)
self.auth_method = config.get("auth_method", "token")
self.secret = config.get("secret", "access_secret")
self.decoder = opuslib_next.Decoder(16000, 1)
self.asr_ws = None
self.forward_task = None
self.conn = None
###################################################################################
# 豆包流式ASR重写父类的方法--开始
###################################################################################
async def open_audio_channels(self, conn):
await super().open_audio_channels(conn)
async with self._session_lock:
# 如果正在重连,等待重连完成
if self.is_reconnecting:
logger.bind(tag=TAG).info("等待当前重连完成...")
await self._session_close_event.wait()
self._session_close_event.clear()
# 如果已有会话未结束,先关闭它
if self._session_started and not self._session_finished:
logger.bind(tag=TAG).warning(
f"发现未关闭的会话 {self._current_session_id},正在关闭..."
)
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
self._session_finished = True
self._session_close_event.set()
# 重置会话状态
self._current_session_id = str(uuid.uuid4())
self._session_started = True
self._session_finished = False
self.is_reconnecting = True
try:
retry_count = 0
while retry_count < self.max_retries:
try:
headers = (
self.token_auth() if self.auth_method == "token" else None
)
self.asr_ws = await websockets.connect(
self.ws_url,
additional_headers=headers,
max_size=1000000000,
ping_interval=None,
ping_timeout=None,
close_timeout=10,
)
# 发送初始化请求
request_params = self.construct_request(
self._current_session_id
)
try:
payload_bytes = str.encode(json.dumps(request_params))
payload_bytes = gzip.compress(payload_bytes)
full_client_request = self.generate_header()
full_client_request.extend(
(len(payload_bytes)).to_bytes(4, "big")
)
full_client_request.extend(payload_bytes)
await self.asr_ws.send(full_client_request)
except Exception as e:
logger.bind(tag=TAG).error(f"发送初始化请求失败: {e}")
raise e
# 等待初始化响应
try:
init_res = await self.asr_ws.recv()
self.parse_response(init_res)
except Exception as e:
logger.bind(tag=TAG).error(f"ASR服务初始化失败: {e}")
raise e
# 启动接收ASR结果的异步任务
with self.thread_lock:
if (
self.asr_thread is None
or not self.asr_thread.is_alive()
):
logger.bind(tag=TAG).info("创建新的ASR监听线程...")
self.asr_thread = threading.Thread(
target=self._start_monitor_asr_response_thread,
daemon=True,
)
self.asr_thread.start()
# 等待一小段时间确保线程启动
await asyncio.sleep(0.1)
if not self.asr_thread.is_alive():
logger.bind(tag=TAG).error("ASR监听线程启动失败")
raise Exception("ASR监听线程启动失败")
logger.bind(tag=TAG).info("ASR监听线程已启动")
return
except websockets.exceptions.WebSocketException as e:
retry_count += 1
if retry_count < self.max_retries:
logger.bind(tag=TAG).warning(
f"WebSocket连接失败,正在进行第{retry_count}次重试: {e}"
)
await asyncio.sleep(self.retry_delay)
else:
logger.bind(tag=TAG).warning(
f"WebSocket连接失败,已达到最大重试次数: {e}"
)
raise
except Exception as e:
logger.bind(tag=TAG).error(f"WebSocket连接发生未知错误: {e}")
raise
finally:
self.is_reconnecting = False
self._session_close_event.set()
async def receive_audio(self, audio, _):
if not isinstance(audio, bytes):
return
try:
# 解码opus得到PCM数据
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)
if self.asr_ws:
await self.asr_ws.send(audio_request)
except Exception as e:
logger.bind(tag=TAG).debug(f"发送音频数据时发生错误: {e}")
###################################################################################
# 豆包流式ASR重写父类的方法--结束
###################################################################################
def construct_request(self, reqid):
req = {
"app": {
"appid": self.appid,
"cluster": self.cluster,
"token": self.access_token,
},
"user": {"uid": self.uid},
"request": {
"reqid": reqid,
"workflow": self.workflow,
"show_utterances": True,
"result_type": self.result_type,
"sequence": 1,
"boosting_table_name": self.boosting_table_name,
"correct_table_name": self.correct_table_name,
},
"audio": {
"format": self.format,
"codec": self.codec,
"rate": self.rate,
"language": self.language,
"bits": self.bits,
"channel": self.channel,
},
}
return req
def token_auth(self):
return {"Authorization": f"Bearer; {self.access_token}"}
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,
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)
header.append((message_type << 4) | message_type_specific_flags)
header.append((serial_method << 4) | compression_type)
header.append(reserved_data)
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,
)
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,
)
def _start_monitor_asr_response_thread(self):
# 初始化链接
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
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}")
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
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
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
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:
"""
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()
@@ -52,6 +52,7 @@ class TTSProviderBase(ABC):
)
self.first_sentence_punctuations = (
"",
"",
"~",
"",
",",