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https://github.com/xinnan-tech/xiaozhi-esp32-server.git
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Merge pull request #2086 from xinnan-tech/ASRPerformanceTester
update:更新流式ASR首词等待时间测试工具
This commit is contained in:
hrz
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import asyncio
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import time
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import json
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import uuid
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import os
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import websockets
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import gzip
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import hmac
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import base64
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import hashlib
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import random
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from urllib import parse
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from tabulate import tabulate
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from config.settings import load_config
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description = "流式ASR首词耗时测试"
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class AccessToken:
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@staticmethod
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def _encode_text(text):
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encoded_text = parse.quote_plus(text)
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return encoded_text.replace("+", "%20").replace("*", "%2A").replace("%7E", "~")
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@staticmethod
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def _encode_dict(dic):
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keys = dic.keys()
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dic_sorted = [(key, dic[key]) for key in sorted(keys)]
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encoded_text = parse.urlencode(dic_sorted)
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return encoded_text.replace("+", "%20").replace("*", "%2A").replace("%7E", "~")
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@staticmethod
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def create_token(access_key_id, access_key_secret):
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parameters = {
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"AccessKeyId": access_key_id,
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"Action": "CreateToken",
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"Format": "JSON",
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"RegionId": "cn-shanghai",
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"SignatureMethod": "HMAC-SHA1",
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"SignatureNonce": str(uuid.uuid1()),
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"SignatureVersion": "1.0",
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"Timestamp": time.strftime("%Y-%m-%dT%H:%M:%SZ", time.gmtime()),
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"Version": "2019-02-28",
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}
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query_string = AccessToken._encode_dict(parameters)
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string_to_sign = (
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"GET" + "&" + AccessToken._encode_text("/") + "&" + AccessToken._encode_text(query_string)
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)
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secreted_string = hmac.new(
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bytes(access_key_secret + "&", encoding="utf-8"),
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bytes(string_to_sign, encoding="utf-8"),
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hashlib.sha1,
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).digest()
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signature = base64.b64encode(secreted_string)
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signature = AccessToken._encode_text(signature)
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full_url = "http://nls-meta.cn-shanghai.aliyuncs.com/?Signature=%s&%s" % (signature, query_string)
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response = requests.get(full_url)
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if response.ok:
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root_obj = response.json()
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if "Token" in root_obj:
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return root_obj["Token"]["Id"], root_obj["Token"]["ExpireTime"]
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return None, None
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class DoubaoStreamASRPerformanceTester:
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def __init__(self):
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self.config = load_config()
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self.test_audio_files = self._load_test_audio_files()
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self.results = []
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def _load_test_audio_files(self):
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"""加载测试用的音频文件"""
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audio_root = os.path.join(os.getcwd(), "config", "assets")
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test_files = []
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if os.path.exists(audio_root):
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for file_name in os.listdir(audio_root):
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if file_name.endswith('.wav') or file_name.endswith('.pcm'):
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with open(os.path.join(audio_root, file_name), 'rb') as f:
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test_files.append(f.read())
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return test_files
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async def test_doubao_stream_asr(self, test_count=5):
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"""测试豆包流式ASR首词响应时间"""
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if not self.test_audio_files:
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print("没有找到测试音频文件")
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return
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asr_config = self.config["ASR"]["DoubaoStreamASR"]
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latencies = []
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for i in range(test_count):
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try:
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ws_url = "wss://openspeech.bytedance.com/api/v3/sauc/bigmodel"
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appid = asr_config["appid"]
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access_token = asr_config["access_token"]
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uid = asr_config.get("uid", "streaming_asr_service")
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start_time = time.time()
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headers = {
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"X-Api-App-Key": appid,
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"X-Api-Access-Key": access_token,
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"X-Api-Resource-Id": "volc.bigasr.sauc.duration",
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"X-Api-Connect-Id": str(uuid.uuid4())
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}
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async with websockets.connect(
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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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) as ws:
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# 发送初始化请求
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request_params = {
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"app": {
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"appid": appid,
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"token": access_token
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},
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"user": {"uid": uid},
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"request": {
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"reqid": str(uuid.uuid4()),
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"workflow": "audio_in,resample,partition,vad,fe,decode,itn,nlu_punctuate",
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"show_utterances": True,
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"result_type": "single",
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"sequence": 1
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},
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"audio": {
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"format": "pcm",
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"codec": "pcm",
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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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"sample_rate": 16000
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}
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}
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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 ws.send(full_client_request)
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init_res = await ws.recv()
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result = self._parse_response(init_res)
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if "code" in result and result["code"] != 1000:
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raise Exception(f"ASR服务初始化失败: {result.get('payload_msg', {}).get('error', '未知错误')}")
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# 发送音频数据
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audio_data = self.test_audio_files[0]
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if audio_data.startswith(b'RIFF'):
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audio_data = audio_data[44:]
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# 直接发送原始音频数据，不进行opus解码
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payload = gzip.compress(audio_data)
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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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await ws.send(audio_request)
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# 等待第一个数据块
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first_chunk = await ws.recv()
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latency = time.time() - start_time
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latencies.append(latency)
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await ws.close()
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except Exception as e:
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print(f"第{i+1}次测试: {str(e)}")
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latencies.append(0)
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return self._calculate_result("豆包流式ASR", latencies, test_count)
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async def test_aliyun_stream_asr(self, test_count=5):
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"""测试阿里云流式ASR首词响应时间"""
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if not self.test_audio_files:
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print("没有找到测试音频文件")
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return
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asr_config = self.config["ASR"]["AliyunStreamASR"]
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latencies = []
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for i in range(test_count):
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try:
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access_key_id = asr_config["access_key_id"]
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access_key_secret = asr_config["access_key_secret"]
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appkey = asr_config["appkey"]
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host = asr_config.get("host", "nls-gateway-cn-shanghai.aliyuncs.com")
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# 获取Token
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token, _ = AccessToken.create_token(access_key_id, access_key_secret)
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if not token:
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raise Exception("无法获取阿里云ASR Token")
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# 确定WebSocket URL
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if "-internal." in host:
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ws_url = f"ws://{host}/ws/v1"
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else:
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ws_url = f"wss://{host}/ws/v1"
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start_time = time.time()
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async with websockets.connect(
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ws_url,
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additional_headers={"X-NLS-Token": token},
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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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) as ws:
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# 发送开始请求
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start_request = {
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"header": {
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"namespace": "SpeechTranscriber",
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"name": "StartTranscription",
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"status": 20000000,
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"message_id": ''.join(random.choices('0123456789abcdef', k=32)),
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"task_id": ''.join(random.choices('0123456789abcdef', k=32)),
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"status_text": "Gateway:SUCCESS:Success.",
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"appkey": appkey
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},
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"payload": {
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"format": "pcm",
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"sample_rate": 16000,
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"enable_intermediate_result": True,
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"enable_punctuation_prediction": True,
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"enable_inverse_text_normalization": True,
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"max_sentence_silence": asr_config.get("max_sentence_silence", 8000),
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"enable_voice_detection": False,
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}
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}
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await ws.send(json.dumps(start_request, ensure_ascii=False))
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# 等待服务器准备
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start_response = await ws.recv()
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response_data = json.loads(start_response)
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if response_data["header"]["name"] != "TranscriptionStarted":
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raise Exception("阿里云ASR服务初始化失败")
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# 发送音频数据
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audio_data = self.test_audio_files[0]
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if audio_data.startswith(b'RIFF'):
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audio_data = audio_data[44:] # 去掉WAV头
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await ws.send(audio_data)
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# 等待第一个结果
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while True:
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response = await ws.recv()
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if isinstance(response, str):
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result = json.loads(response)
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if result["header"]["name"] == "TranscriptionResultChanged":
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latency = time.time() - start_time
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latencies.append(latency)
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break
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elif result["header"]["name"] == "TaskFailed":
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raise Exception(f"阿里云ASR识别失败: {result.get('payload', {}).get('error_info', '未知错误')}")
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# 发送停止请求
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stop_msg = {
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"header": {
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"namespace": "SpeechTranscriber",
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"name": "StopTranscription",
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"status": 20000000,
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"message_id": ''.join(random.choices('0123456789abcdef', k=32)),
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"status_text": "Client:Stop",
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"appkey": appkey
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}
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}
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await ws.send(json.dumps(stop_msg, ensure_ascii=False))
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await ws.close()
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except Exception as e:
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print(f"第{i+1}次测试: {str(e)}")
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latencies.append(0)
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return self._calculate_result("阿里云流式ASR", latencies, test_count)
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def _generate_header(self):
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"""生成请求头"""
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header = bytearray()
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header.append((0x01 << 4) | 0x01)
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header.append((0x01 << 4) | 0x00)
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header.append((0x01 << 4) | 0x01)
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header.append(0x00)
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return header
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def _generate_audio_default_header(self):
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"""生成音频请求头"""
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return self._generate_header()
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def _parse_response(self, res: bytes) -> dict:
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"""解析响应"""
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try:
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if len(res) < 4:
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return {"error": "响应数据长度不足"}
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header = res[:4]
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message_type = header[1] >> 4
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if message_type == 0x0F:
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code = int.from_bytes(res[4:8], "big", signed=False)
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msg_length = int.from_bytes(res[8:12], "big", signed=False)
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error_msg = json.loads(res[12:].decode("utf-8"))
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return {
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"code": code,
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"msg_length": msg_length,
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"payload_msg": error_msg
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}
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try:
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json_data = res[12:].decode("utf-8")
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return {"payload_msg": json.loads(json_data)}
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except (UnicodeDecodeError, json.JSONDecodeError):
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return {"error": "JSON解析失败"}
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except Exception:
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return {"error": "解析响应失败"}
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def _calculate_result(self, service_name, latencies, test_count):
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"""计算结果"""
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valid_latencies = [l for l in latencies if l > 0]
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if valid_latencies:
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avg_latency = sum(valid_latencies) / len(valid_latencies)
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status = f"成功（{len(valid_latencies)}/{test_count}次有效）"
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else:
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avg_latency = 0
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status = "失败: 所有测试均失败"
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return {"name": service_name, "latency": avg_latency, "status": status}
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def _print_results(self, test_count):
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"""打印测试结果"""
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if not self.results:
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print("没有有效的ASR测试结果")
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return
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print(f"\n{'='*60}")
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print("流式ASR首词响应时间测试结果")
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print(f"{'='*60}")
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print(f"测试次数: 每个ASR服务测试 {test_count} 次")
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# 排序结果：成功优先，按延迟升序
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success_results = sorted(
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[r for r in self.results if "成功" in r["status"]],
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key=lambda x: x["latency"]
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)
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failed_results = [r for r in self.results if "成功" not in r["status"]]
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table_data = [
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[r["name"], f"{r['latency']:.3f}", r["status"]]
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for r in success_results + failed_results
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]
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print(tabulate(table_data, headers=["ASR服务", "首词延迟(秒)", "状态"], tablefmt="grid"))
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print("\n测试说明：测量从发送请求到接收第一个识别结果的时间，取多次测试平均值")
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print("- 超时控制: 单个请求最大等待时间为10秒")
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print("- 错误处理: 无法连接和超时的列为网络错误")
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print("- 排序规则: 按平均耗时从快到慢排序")
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async def run(self, test_count=5):
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"""执行测试"""
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print(f"开始流式ASR首词响应时间测试...")
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print(f"每个ASR服务测试次数: {test_count}次")
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if not self.config.get("ASR"):
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print("配置文件中未找到ASR配置")
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return
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# 测试每种ASR服务
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self.results = []
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# 测试豆包ASR
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if self.config["ASR"].get("DoubaoStreamASR"):
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result = await self.test_doubao_stream_asr(test_count)
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self.results.append(result)
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else:
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print("配置文件中未找到豆包流式ASR配置，跳过测试")
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# 测试阿里云ASR
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if self.config["ASR"].get("AliyunStreamASR"):
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result = await self.test_aliyun_stream_asr(test_count)
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self.results.append(result)
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else:
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print("配置文件中未找到阿里云流式ASR配置，跳过测试")
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# 打印结果
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self._print_results(test_count)
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async def main():
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import argparse
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parser = argparse.ArgumentParser(description="流式ASR首词响应时间测试工具")
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parser.add_argument("--count", type=int, default=5, help="测试次数")
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args = parser.parse_args()
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await DoubaoStreamASRPerformanceTester().run(args.count)
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if __name__ == "__main__":
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import os
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import gzip
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import opuslib_next
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asyncio.run(main())
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