[说话人认证]编写SPKFunctor，使用cam++提取embs，用余弦相似度判断得分。

src/core/__init__.py

@@ -0,0 +1,3 @@
+from .model_loader import ModelLoader
+
+__all__ = ["ModelLoader"]

src/functor/spk_functor.py

@@ -4,16 +4,41 @@ SpkFunctor
 """
 
 from src.functor.base import BaseFunctor
-from src.models import AudioBinary_Config, VAD_Functor_result
-from typing import Callable, List
+from src.models import AudioBinary_Config, VAD_Functor_result, SpeakerCreate
+from typing import Callable, List, Dict
 from queue import Queue, Empty
+import json
+import torch
 import threading
+import numpy
 
 # 日志
 from src.utils.logger import get_module_logger
 
 logger = get_module_logger(__name__)
-
+from modelscope.pipelines import pipeline
+# sv_pipeline = pipeline(
+#     task='speaker-verification',
+#     model='iic/speech_campplus_sv_zh-cn_16k-common',
+#     model_revision='v1.0.0'
+# )
+# speaker1_a_wav = 'https://modelscope.cn/api/v1/models/damo/speech_campplus_sv_zh-cn_16k-common/repo?Revision=master&FilePath=examples/speaker1_a_cn_16k.wav'
+# speaker1_b_wav = 'https://modelscope.cn/api/v1/models/damo/speech_campplus_sv_zh-cn_16k-common/repo?Revision=master&FilePath=examples/speaker1_b_cn_16k.wav'
+# speaker2_a_wav = 'https://modelscope.cn/api/v1/models/damo/speech_campplus_sv_zh-cn_16k-common/repo?Revision=master&FilePath=examples/speaker2_a_cn_16k.wav'
+# # 相同说话人语音
+# result = sv_pipeline([speaker1_a_wav, speaker1_b_wav])
+# print(result)
+# # 不同说话人语音
+# result = sv_pipeline([speaker1_a_wav, speaker2_a_wav])
+# print(result)
+# # 可以自定义得分阈值来进行识别，阈值越高，判定为同一人的条件越严格
+# result = sv_pipeline([speaker1_a_wav, speaker1_a_wav], thr=0.6)
+# print(result)
+# # 可以传入output_emb参数，输出结果中就会包含提取到的说话人embedding
+# result = sv_pipeline([speaker1_a_wav, speaker2_a_wav], output_emb=True)
+# print(result['embs'], result['outputs'])
+# # 可以传入save_dir参数，提取到的说话人embedding会存储在save_dir目录中
+# result = sv_pipeline([speaker1_a_wav, speaker2_a_wav], save_dir='savePath/')
 
 class SPKFunctor(BaseFunctor):
     """
@@ -26,15 +51,84 @@ class SPKFunctor(BaseFunctor):
 
     使用stop()停止线程, 但需要等待input_queue为空
     """
+    class speaker_verify:
+        """
+        说话人验证
+        """
+        def __init__(self, *args, **kwargs):
+            self._spk_data: List[SpeakerCreate] = []
 
+        def add_speaker(self, speaker: SpeakerCreate) -> None:
+            self._spk_data.append(speaker)
+
+        def verify(self, emb: numpy.ndarray) -> Dict:
+            # 将输入的numpy embedding转换为tensor
+            input_emb_tensor = torch.from_numpy(emb).unsqueeze(0)
+            
+            results = []
+            for speaker in self._spk_data:
+                if not speaker.speaker_embs:
+                    continue
+                # 将列表中存储的embedding转换为tensor
+                speaker_emb_tensor = torch.tensor(speaker.speaker_embs).unsqueeze(0)
+                
+                # 计算余弦相似度
+                score = torch.nn.functional.cosine_similarity(input_emb_tensor, speaker_emb_tensor).item()
+
+                result = {
+                    "score": score,
+                    "speaker_id": speaker.speaker_id,
+                    "speaker_name": speaker.speaker_name,
+                    "speaker_description": speaker.speaker_description
+                }
+                results.append(result)
+
+            if not results:
+                return {
+                    "speaker_id": "unknown",
+                    "speaker_name": "Unknown",
+                    "speaker_description": "No registered speakers to verify against.",
+                    "score": 0.0,
+                    "results": []
+                }
+
+            results.sort(key=lambda x: x['score'], reverse=True)
+            best_match = results[0]
+            return {
+                "speaker_id": best_match['speaker_id'],
+                "speaker_name": best_match['speaker_name'],
+                "speaker_description": best_match['speaker_description'],
+                "score": best_match['score'],
+                "results": results
+            }
+            
     def __init__(self) -> None:
         super().__init__()
         # 资源与配置
+        self._spk_verify = self.speaker_verify()
+        self._sv_pipeline = pipeline(
+            task='speaker-verification',
+            model='iic/speech_campplus_sv_zh-cn_16k-common',
+            model_revision='v1.0.0'
+        )
         self._model: dict = {}  # 模型
         self._callback: List[Callable] = []  # 回调函数
         self._input_queue: Queue = None  # 输入队列
         self._audio_config: AudioBinary_Config = None  # 音频配置
 
+    def load_spk_data_local(
+        self,
+        spk_data_path: str = 'data/speakers.json',
+        ) -> None:
+        """
+        加载本地说话人数据
+        """
+        with open(spk_data_path, 'r') as f:
+            spk_data = json.load(f)
+            for spk in spk_data:
+                self._spk_verify.add_speaker(SpeakerCreate(**spk))
+        
+
     def reset_cache(self) -> None:
         """
         重置缓存, 用于任务完成后清理缓存数据, 准备下次任务
@@ -60,6 +154,7 @@ class SPKFunctor(BaseFunctor):
         self._audio_config = audio_config
         logger.debug("SpkFunctor设置音频配置: %s", self._audio_config)
 
+
     def add_callback(self, callback: Callable) -> None:
         """
         向自身的_callback: List[Callable]回调函数列表中添加回调函数
@@ -84,7 +179,11 @@ class SPKFunctor(BaseFunctor):
         #     input=binary_data,
         #     chunk_size=self._audio_config.chunk_size,
         # )
-        result = [{"result": "spk1", "score": {"spk1": 0.9, "spk2": 0.3}}]
+        sv_result = self._sv_pipeline([binary_data], output_emb=True)
+        embs = sv_result['embs'][0]
+
+        result = self._spk_verify.verify(embs)
+        
         self._do_callback(result)
 
     def _run(self) -> None:

src/models/__init__.py

@@ -1,6 +1,8 @@
 from .audio import AudioBinary_Config, AudioBinary_data_list, _AudioBinary_data
 from .vad import VAD_Functor_result
 
+from .spk import SpeakerCreate, SpeakerResponse
+
 __all__ = [
     "AudioBinary_Config",
     "AudioBinary_data_list",

src/models/spk.py

@@ -0,0 +1,68 @@
+"""
+src/schemas/speaker.py
+------------------------
+此模块定义与说话人（speakers）表对应的 Pydantic 模型，用于 API 数据验证和序列化。
+
+模型说明：
+- SpeakerBase：定义说话人的基础字段，包括姓名与描述。
+- SpeakerCreate：用于创建说话人时的数据验证，直接继承 SpeakerBase。
+- SpeakerUpdate：用于更新说话人信息时，所有字段均为可选。
+- SpeakerResponse：返回给客户端时使用，包含数据库生成的字段（如 speaker_id、created_at 等）。
+"""
+
+from datetime import datetime
+from typing import Optional, List
+from uuid import UUID
+from pydantic import BaseModel, Field
+from .base import BaseSchema
+
+# 基础模型，定义说话人的核心属性
+class SpeakerBase(BaseSchema):
+    speaker_id: UUID = Field(
+        ..., 
+        description="说话人唯一标识符"
+    )
+    speaker_name: str = Field(
+        ..., 
+        max_length=255, 
+        description="说话人姓名，必填，最多255字符"
+    )
+    speaker_description: Optional[str] = Field(
+        None, 
+        description="说话人描述信息，可选"
+    )
+    speaker_embs: Optional[List[float]] = Field(
+        None, 
+        description="说话人embedding，可选"
+    )
+
+# 用于创建说话人时的数据模型，直接继承 SpeakerBase
+class SpeakerCreate(SpeakerBase):
+    pass
+
+# 用于更新说话人时的数据模型，所有字段都是可选的
+class SpeakerUpdate(BaseModel):
+    speaker_name: Optional[str] = Field(
+        None, 
+        max_length=255, 
+        description="说话人姓名"
+    )
+    speaker_description: Optional[str] = Field(
+        None, 
+        description="说话人描述信息"
+    )
+
+# 用于 API 返回的说话人响应模型，扩展基础模型以包含数据库生成的字段
+class SpeakerResponse(SpeakerBase):
+    speaker_id: UUID = Field(
+        ..., 
+        description="说话人的唯一标识符（UUID）"
+    )
+    created_at: datetime = Field(
+        ..., 
+        description="记录创建时间"
+    )
+    updated_at: datetime = Field(
+        ..., 
+        description="最近更新时间"
+    )

src/websockets/endpoint/__init__.py

@@ -0,0 +1,3 @@
+from .asr_endpoint import router as asr_router
+
+__all__ = ["asr_router"]

src/websockets/endpoint/asr_endpoint.py

@@ -0,0 +1,32 @@
+"""
+ -*- coding: utf-8 -*-
+ 此模块是ASR的websocket端点, 使用FastAPI的websocket端点
+"""
+
+from fastapi import WebSocket, APIRouter
+
+router = APIRouter()
+
+from src.runner.ASRRunner import ASRRunner
+
+ASRRunner_instance = ASRRunner()
+
+from src.core import ModelLoader
+model_loader = ModelLoader()
+args = {
+    "asr_model": "paraformer-zh",
+    "asr_model_revision": "v2.0.4",
+    "vad_model": "fsmn-vad",
+    "vad_model_revision": "v2.0.4",
+    "spk_model": "cam++",
+    "spk_model_revision": "v2.0.2",
+    "audio_update": False,
+}
+models = model_loader.load_models(args)
+
+@router.websocket("/asr_full")
+async def asr_endpoint(websocket: WebSocket):
+    await websocket.accept()
+    while True:
+        data = await websocket.receive_text()
+        print(data)

src/websockets/router.py

@@ -0,0 +1,3 @@
+from endpoint import asr_router
+
+__all__ = ["asr_router"]

test_main.py

@@ -5,7 +5,7 @@
 
 from tests.pipeline.asr_test import test_asr_pipeline
 from src.utils.logger import get_module_logger, setup_root_logger
-from tests.runner.stt_runner_test import test_asr_runner
+from tests.runner.asr_runner_test import test_asr_runner
 
 setup_root_logger(level="INFO", log_file="logs/test_main.log")
 logger = get_module_logger(__name__)

tests/spkverify_use.py

@@ -19,5 +19,7 @@ print(result)
 # 可以传入output_emb参数，输出结果中就会包含提取到的说话人embedding
 result = sv_pipeline([speaker1_a_wav, speaker2_a_wav], output_emb=True)
 print(result['embs'], result['outputs'])
+# result1 = sv_pipeline([speaker1_a_wav], output_emb=True)
+# print(result1['embs'], result1['outputs'])
 # 可以传入save_dir参数，提取到的说话人embedding会存储在save_dir目录中
 result = sv_pipeline([speaker1_a_wav, speaker2_a_wav], save_dir='savePath/')