commit 31017e1ec483e7d83c0d42070d0585cf6348e7fc
Author: bing <2524698668@qq.com>
Date:   Sat May 11 22:02:52 2024 +0800

    utils for punctuation and emotion and speaker ver

diff --git a/takway/stt/__init__.py b/takway/stt/__init__.py
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@@ -0,0 +1 @@
+from .base_stt import *
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diff --git a/takway/stt/__pycache__/emotion_utils.cpython-39.pyc b/takway/stt/__pycache__/emotion_utils.cpython-39.pyc
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diff --git a/takway/stt/__pycache__/funasr_utils.cpython-39.pyc b/takway/stt/__pycache__/funasr_utils.cpython-39.pyc
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diff --git a/takway/stt/__pycache__/modified_funasr.cpython-39.pyc b/takway/stt/__pycache__/modified_funasr.cpython-39.pyc
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diff --git a/takway/stt/__pycache__/punctuation_utils.cpython-39.pyc b/takway/stt/__pycache__/punctuation_utils.cpython-39.pyc
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diff --git a/takway/stt/__pycache__/speaker_ver_utils.cpython-39.pyc b/takway/stt/__pycache__/speaker_ver_utils.cpython-39.pyc
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diff --git a/takway/stt/base_stt.py b/takway/stt/base_stt.py
new file mode 100644
index 0000000..4763446
--- /dev/null
+++ b/takway/stt/base_stt.py
@@ -0,0 +1,65 @@
+#!/usr/bin/env python3
+# -*- coding: utf-8 -*-
+import json
+import wave
+import io
+import os
+import logging
+from ..common_utils import decode_str2bytes
+
+class STTBase:
+    def __init__(self, RATE=16000, cfg_path=None, debug=False):
+        self.RATE = RATE
+        self.debug = debug
+        self.asr_cfg = self.parse_json(cfg_path)
+        
+    def parse_json(self, cfg_path):
+        cfg = None
+        self.hotwords = None
+        if cfg_path is not None:
+            with open(cfg_path, 'r', encoding='utf-8') as f:
+                cfg = json.load(f)
+            self.hotwords = cfg.get('hot_words', None)
+            logging.info(f"load STT config file: {cfg_path}")
+            logging.info(f"Hot words: {self.hotwords}")
+        else:
+            logging.warning("No STT config file provided, using default config.")
+        return cfg
+
+    def add_hotword(self, hotword):
+        """add hotword to list"""
+        if self.hotwords is None:
+            self.hotwords = ""
+        if isinstance(hotword, str):
+            self.hotwords = self.hotwords + " " + "hotword"
+        elif isinstance(hotword, (list, tuple)):
+            # 将hotwords转换为str，并用空格隔开
+            self.hotwords = self.hotwords + " " + " ".join(hotword)
+        else:
+            raise TypeError("hotword must be str or list")
+
+    def check_audio_type(self, audio_data):
+        """check audio data type and convert it to bytes if necessary."""
+        if isinstance(audio_data, bytes):
+            pass
+        elif isinstance(audio_data, list):
+            audio_data = b''.join(audio_data)
+        elif isinstance(audio_data, str):
+            audio_data = decode_str2bytes(audio_data)
+        elif isinstance(audio_data, io.BytesIO):
+            wf = wave.open(audio_data, 'rb')
+            audio_data = wf.readframes(wf.getnframes())
+        else:
+            raise TypeError(f"audio_data must be bytes, str or io.BytesIO, but got {type(audio_data)}")
+        return audio_data
+    
+    def text_postprecess(self, result, data_id='text'):
+        """postprecess recognized result."""
+        text = result[data_id]
+        if isinstance(text, list):
+            text = ''.join(text)
+        return text.replace(' ', '')
+
+    def recognize(self, audio_data, queue=None):
+        """recognize audio data to text"""
+        pass
diff --git a/takway/stt/emotion_utils.py b/takway/stt/emotion_utils.py
new file mode 100644
index 0000000..8d38423
--- /dev/null
+++ b/takway/stt/emotion_utils.py
@@ -0,0 +1,142 @@
+import io
+import numpy as np
+import base64
+import wave
+from funasr import AutoModel
+import time
+"""
+    Base模型
+        不能进行情绪分类,只能用作特征提取
+"""
+FUNASRBASE = {
+    "model_type": "funasr",
+    "model_path": "iic/emotion2vec_base",
+    "model_revision": "v2.0.4"
+}
+
+
+"""
+    Finetune模型
+        输出分类结果
+"""
+FUNASRFINETUNE = {
+    "model_type": "funasr",
+    "model_path": "iic/emotion2vec_base_finetuned"
+}
+
+def decode_str2bytes(data):
+    # 将Base64编码的字节串解码为字节串
+    if data is None:
+        return None
+    return base64.b64decode(data.encode('utf-8'))
+
+class Emotion:
+    def __init__(self,
+                 model_type="funasr",
+                 model_path="iic/emotion2vec_base",
+                 device="cuda",
+                 model_revision="v2.0.4",
+                 **kwargs):
+        
+        self.model_type = model_type
+        self.initialize(model_type, model_path, device, model_revision, **kwargs)
+    
+    # 初始化模型
+    def initialize(self,
+                   model_type,
+                   model_path,
+                   device,
+                   model_revision,
+                   **kwargs):
+        if model_type == "funasr":
+            self.emotion_model = AutoModel(model=model_path, device=device, model_revision=model_revision, **kwargs)
+        else:
+            raise NotImplementedError(f"unsupported model type [{model_type}]. only [funasr] expected.")
+
+    # 检查输入类型
+    def check_audio_type(self,
+                         audio_data):
+        """check audio data type and convert it to bytes if necessary."""
+        if isinstance(audio_data, bytes):
+            pass
+        elif isinstance(audio_data, list):
+            audio_data = b''.join(audio_data)
+        elif isinstance(audio_data, str):
+            audio_data = decode_str2bytes(audio_data)
+        elif isinstance(audio_data, io.BytesIO):
+            wf = wave.open(audio_data, 'rb')
+            audio_data = wf.readframes(wf.getnframes())
+        elif isinstance(audio_data, np.ndarray):
+            pass
+        else:
+            raise TypeError(f"audio_data must be bytes, list, str, \
+                io.BytesIO or numpy array, but got {type(audio_data)}")
+        
+        if isinstance(audio_data, bytes):
+            audio_data = np.frombuffer(audio_data, dtype=np.int16)
+        elif isinstance(audio_data, np.ndarray):
+            if audio_data.dtype != np.int16:
+                audio_data = audio_data.astype(np.int16)
+        else:
+            raise TypeError(f"audio_data must be bytes or numpy array, but got {type(audio_data)}")
+        
+        # 输入类型必须是float32
+        if isinstance(audio_data, np.ndarray):
+            audio_data = audio_data.astype(np.float32)
+        else:
+            raise TypeError(f"audio_data must be numpy array, but got {type(audio_data)}")
+        return audio_data
+    
+    def process(self,
+                audio_data,
+                granularity="utterance",
+                extract_embedding=False,
+                output_dir=None,
+                only_score=True):
+        """
+            audio_data: only float32 expected beacause layernorm
+            extract_embedding: save embedding if true
+            output_dir: save path for embedding
+            only_Score: only return lables & scores if true
+        """
+        audio_data = self.check_audio_type(audio_data)
+        if self.model_type == 'funasr':
+            result = self.emotion_model.generate(audio_data, output_dir=output_dir, granularity=granularity, extract_embedding=extract_embedding)
+        else:
+            pass
+
+        # 只保留 lables 和 scores
+        if only_score:
+            maintain_key = ["labels", "scores"]
+            for res in result:
+                keys_to_remove = [k for k in res.keys() if k not in maintain_key]
+                for k in keys_to_remove:
+                    res.pop(k)
+        return result[0]
+
+# only for test
+def load_audio_file(wav_file):
+    with wave.open(wav_file, 'rb') as wf:
+        params = wf.getparams()
+        frames = wf.readframes(params.nframes)
+        print("Audio file loaded.")
+        # Audio Parameters
+        # print("Channels:", params.nchannels)
+        # print("Sample width:", params.sampwidth)
+        # print("Frame rate:", params.framerate)
+        # print("Number of frames:", params.nframes)
+        # print("Compression type:", params.comptype)
+    return frames
+
+if __name__ == "__main__":
+    inputs = r".\example\test.wav"
+    inputs = load_audio_file(inputs)
+    device = "cuda"
+    # FUNASRBASE.update({"device": device})
+    FUNASRFINETUNE.update({"deivce": device})
+    emotion_model = Emotion(**FUNASRFINETUNE)
+    s = time.time()
+    result = emotion_model.process(inputs)
+    t = time.time()
+    print(t - s)
+    print(result)
\ No newline at end of file
diff --git a/takway/stt/funasr_utils.py b/takway/stt/funasr_utils.py
new file mode 100644
index 0000000..92eb41b
--- /dev/null
+++ b/takway/stt/funasr_utils.py
@@ -0,0 +1,186 @@
+#!/usr/bin/env python3
+# -*- coding: utf-8 -*-
+# ####################################################### #
+# FunAutoSpeechRecognizer: https://github.com/alibaba-damo-academy/FunASR
+# ####################################################### #
+import io
+import time
+import numpy as np
+from takway.common_utils import decode_str2bytes
+from funasr import AutoModel
+
+from takway.stt.base_stt import STTBase
+
+class FunAutoSpeechRecognizer(STTBase):
+    def __init__(self, 
+                 model_path="paraformer-zh-streaming", 
+                 device="cuda", 
+                 RATE=16000, 
+                 cfg_path=None, 
+                 debug=False, 
+                 chunk_ms=480, 
+                 encoder_chunk_look_back=4, 
+                 decoder_chunk_look_back=1, 
+                 **kwargs):
+        super().__init__(RATE=RATE, cfg_path=cfg_path, debug=debug)
+        
+        self.asr_model = AutoModel(model=model_path, device=device, **kwargs)
+        
+        self.encoder_chunk_look_back = encoder_chunk_look_back #number of chunks to lookback for encoder self-attention
+        self.decoder_chunk_look_back = decoder_chunk_look_back #number of encoder chunks to lookback for decoder cross-attention
+        
+        #[0, 8, 4] 480ms, [0, 10, 5] 600ms
+        if chunk_ms == 480:
+            self.chunk_size = [0, 8, 4] 
+        elif chunk_ms == 600:
+            self.chunk_size = [0, 10, 5]
+        else:
+            raise ValueError("`chunk_ms` should be 480 or 600, and type is int.")
+        self.chunk_partial_size = self.chunk_size[1] * 960 
+        self.audio_cache = None
+        self.asr_cache = {}
+        
+        
+        
+        self._init_asr()
+        
+    def check_audio_type(self, audio_data):
+        """check audio data type and convert it to bytes if necessary."""
+        if isinstance(audio_data, bytes):
+            pass
+        elif isinstance(audio_data, list):
+            audio_data = b''.join(audio_data)
+        elif isinstance(audio_data, str):
+            audio_data = decode_str2bytes(audio_data)
+        elif isinstance(audio_data, io.BytesIO):
+            wf = wave.open(audio_data, 'rb')
+            audio_data = wf.readframes(wf.getnframes())
+        elif isinstance(audio_data, np.ndarray):
+            pass
+        else:
+            raise TypeError(f"audio_data must be bytes, list, str, \
+                io.BytesIO or numpy array, but got {type(audio_data)}")
+        
+        if isinstance(audio_data, bytes):
+            audio_data = np.frombuffer(audio_data, dtype=np.int16)
+        elif isinstance(audio_data, np.ndarray):
+            if audio_data.dtype != np.int16:
+                audio_data = audio_data.astype(np.int16)
+        else:
+            raise TypeError(f"audio_data must be bytes or numpy array, but got {type(audio_data)}")
+        return audio_data
+    
+    def _init_asr(self):
+        # 随机初始化一段音频数据
+        init_audio_data = np.random.randint(-32768, 32767, size=self.chunk_partial_size, dtype=np.int16)
+        self.asr_model.generate(input=init_audio_data, cache=self.asr_cache, is_final=False, chunk_size=self.chunk_size, encoder_chunk_look_back=self.encoder_chunk_look_back, decoder_chunk_look_back=self.decoder_chunk_look_back)
+        self.audio_cache = None
+        self.asr_cache = {}
+        print("init ASR model done.")
+    
+    def recognize(self, audio_data):
+        """recognize audio data to text"""
+        audio_data = self.check_audio_type(audio_data)
+        result = self.asr_model.generate(input=audio_data, 
+                     batch_size_s=300, 
+                     hotword=self.hotwords)
+        
+        # print(result)
+        text = ''
+        for res in result:
+            text += res['text']
+        return text
+    
+    def streaming_recognize(self, 
+                            audio_data, 
+                            is_end=False, 
+                            auto_det_end=False):
+        """recognize partial result
+        
+        Args:
+            audio_data: bytes or numpy array, partial audio data
+            is_end: bool, whether the audio data is the end of a sentence
+            auto_det_end: bool, whether to automatically detect the end of a audio data
+        """
+        text_dict = dict(text=[], is_end=is_end)
+        
+        audio_data = self.check_audio_type(audio_data)
+        if self.audio_cache is None:
+            self.audio_cache = audio_data
+        else:
+            # print(f"audio_data: {audio_data.shape}, audio_cache: {self.audio_cache.shape}")
+            if self.audio_cache.shape[0] > 0:
+                self.audio_cache = np.concatenate([self.audio_cache, audio_data], axis=0)
+        
+        if not is_end and self.audio_cache.shape[0] < self.chunk_partial_size:
+            return text_dict
+        
+        total_chunk_num = int((len(self.audio_cache)-1)/self.chunk_partial_size)
+        
+        if is_end:
+            # if the audio data is the end of a sentence, \
+            # we need to add one more chunk to the end to \
+            # ensure the end of the sentence is recognized correctly.
+            auto_det_end = True
+        
+        if auto_det_end:
+            total_chunk_num += 1
+
+        # print(f"chunk_size: {self.chunk_size}, chunk_stride: {self.chunk_partial_size}, total_chunk_num: {total_chunk_num}, len: {len(self.audio_cache)}")
+        end_idx = None
+        for i in range(total_chunk_num):
+            if auto_det_end:
+                is_end = i == total_chunk_num - 1
+            start_idx = i*self.chunk_partial_size
+            if auto_det_end:
+                end_idx = (i+1)*self.chunk_partial_size if i < total_chunk_num-1 else -1
+            else:
+                end_idx = (i+1)*self.chunk_partial_size if i < total_chunk_num else -1
+            # print(f"cut part: {start_idx}:{end_idx}, is_end: {is_end}, i: {i}, total_chunk_num: {total_chunk_num}")
+            # t_stamp = time.time()
+            
+            speech_chunk = self.audio_cache[start_idx:end_idx]
+
+            # TODO: exceptions processes
+            try:
+                res = self.asr_model.generate(input=speech_chunk, cache=self.asr_cache, is_final=is_end, chunk_size=self.chunk_size, encoder_chunk_look_back=self.encoder_chunk_look_back, decoder_chunk_look_back=self.decoder_chunk_look_back)
+            except ValueError as e:
+                print(f"ValueError: {e}")
+                continue
+            text_dict['text'].append(self.text_postprecess(res[0], data_id='text'))
+            # print(f"each chunk time: {time.time()-t_stamp}")
+            
+        if is_end:
+            self.audio_cache = None
+            self.asr_cache = {}
+        else:
+            if end_idx:
+                self.audio_cache = self.audio_cache[end_idx:] # cut the processed part from audio_cache
+        text_dict['is_end'] = is_end
+        
+        # print(f"text_dict: {text_dict}")
+        return text_dict
+
+    
+
+if __name__ == '__main__':
+    from takway.audio_utils import BaseAudio
+    rec = BaseAudio(input=True, CHUNK=3840)
+    
+    # return_type = 'bytes'
+    file_path = 'my_recording.wav'
+    data = rec.load_audio_file(file_path)
+        
+    asr = FunAutoSpeechRecognizer()
+    
+    # asr.recognize(data)
+    total_chunk_num = int((len(data)-1)/rec.CHUNK+1)
+    print(f"total_chunk_num: {total_chunk_num}")
+    for i in range(total_chunk_num):
+        is_end = i == total_chunk_num - 1
+        speech_chunk = data[i*rec.CHUNK:(i+1)*rec.CHUNK]
+        text_dict = asr.streaming_recognize(speech_chunk, is_end)
+    '''
+    asr.streaming_recognize(data, auto_det_end=True)
+    '''
+    
\ No newline at end of file
diff --git a/takway/stt/modified_funasr.py b/takway/stt/modified_funasr.py
new file mode 100644
index 0000000..5628aad
--- /dev/null
+++ b/takway/stt/modified_funasr.py
@@ -0,0 +1,168 @@
+from takway.stt.funasr_utils import FunAutoSpeechRecognizer
+from takway.stt.punctuation_utils import CTTRANSFORMER, Punctuation
+from takway.stt.emotion_utils import FUNASRFINETUNE, Emotion
+from takway.stt.speaker_ver_utils import ERES2NETV2, DEFALUT_SAVE_PATH, speaker_verfication
+import os
+import pdb
+import numpy as np
+class ModifiedRecognizer(FunAutoSpeechRecognizer):
+    def __init__(self, 
+                 use_punct=True, 
+                 use_emotion=False,
+                 use_speaker_ver=True):
+        super().__init__(
+                 model_path="paraformer-zh-streaming", 
+                 device="cuda", 
+                 RATE=16000, 
+                 cfg_path=None, 
+                 debug=False, 
+                 chunk_ms=480, 
+                 encoder_chunk_look_back=4, 
+                 decoder_chunk_look_back=1)
+        self.use_punct = use_punct
+        self.use_emotion = use_emotion
+        self.use_speaker_ver = use_speaker_ver
+
+        if use_punct:
+            self.puctuation_model = Punctuation(**CTTRANSFORMER)
+        if use_emotion:
+            self.emotion_model = Emotion(**FUNASRFINETUNE)
+        if use_speaker_ver:
+            self.speaker_ver_model = speaker_verfication(**ERES2NETV2)
+
+    def initialize_speaker(self, speaker_1_wav):
+        if not self.use_speaker_ver:
+            raise NotImplementedError("no access")
+        if speaker_1_wav.endswith(".npy"):
+            self.save_speaker_path = speaker_1_wav
+        elif speaker_1_wav.endswith('.wav'):
+            self.save_speaker_path = os.path.join(DEFALUT_SAVE_PATH, 
+                                          os.path.basename(speaker_1_wav).replace(".wav", ".npy"))
+            # self.save_speaker_path = DEFALUT_SAVE_PATH
+            self.speaker_ver_model.wav2embeddings(speaker_1_wav, self.save_speaker_path)
+        else:
+            raise TypeError("only support [.npy] or [.wav].")
+            
+
+    def speaker_ver(self, speaker_2_wav):
+        if not self.use_speaker_ver:
+            raise NotImplementedError("no access")
+        if not hasattr(self, "save_speaker_path"):
+            raise NotImplementedError("please initialize speaker first")
+        # pdb.set_trace()
+        return self.speaker_ver_model.verfication(base_emb=self.save_speaker_path,
+                                           speaker_2_wav=speaker_2_wav) == 'yes'
+        
+
+    def recognize(self, audio_data):
+        audio_data = self.check_audio_type(audio_data)
+
+        if self.use_speaker_ver:
+            if self.speaker_ver_model.verfication(self.save_speaker_path, 
+                                                  speaker_2_wav=audio_data) == 'no':
+                return "Other People"
+
+        result = self.asr_model.generate(input=audio_data,
+                                         batch_size_s=300,
+                                         hotword=self.hotwords)
+        text = ''
+        for res in result:
+            text += res['text']
+        if self.use_punct:
+            text = self.puctuation_model.process(text+'#', append_period=False).replace('#', '')
+
+        return text
+    
+    def recognize_emotion(self, audio_data):
+        audio_data = self.check_audio_type(audio_data)
+
+        if self.use_speaker_ver:
+            if self.speaker_ver_model.verfication(self.save_speaker_path, 
+                                                  speaker_2_wav=audio_data) == 'no':
+                return "Other People"
+            
+        if self.use_emotion:
+            return self.emotion_model.process(audio_data)
+        else:
+            raise NotImplementedError("no access")
+
+    def streaming_recognize(self, audio_data, is_end=False, auto_det_end=False):
+        """recognize partial result
+        
+        Args:
+            audio_data: bytes or numpy array, partial audio data
+            is_end: bool, whether the audio data is the end of a sentence
+            auto_det_end: bool, whether to automatically detect the end of a audio data
+        """
+        audio_data = self.check_audio_type(audio_data)
+
+        if self.use_speaker_ver:
+            if self.speaker_ver_model.verfication(self.save_speaker_path, 
+                                                  speaker_2_wav=audio_data) == 'no':
+                return "Other People"
+        
+        text_dict = dict(text=[], is_end=is_end)
+        
+        if self.audio_cache is None:
+            self.audio_cache = audio_data
+        else:
+            # print(f"audio_data: {audio_data.shape}, audio_cache: {self.audio_cache.shape}")
+            if self.audio_cache.shape[0] > 0:
+                self.audio_cache = np.concatenate([self.audio_cache, audio_data], axis=0)
+        
+        if not is_end and self.audio_cache.shape[0] < self.chunk_partial_size:
+            return text_dict
+        
+        total_chunk_num = int((len(self.audio_cache)-1)/self.chunk_partial_size)
+        
+        if is_end:
+            # if the audio data is the end of a sentence, \
+            # we need to add one more chunk to the end to \
+            # ensure the end of the sentence is recognized correctly.
+            auto_det_end = True
+        
+        if auto_det_end:
+            total_chunk_num += 1
+
+        # print(f"chunk_size: {self.chunk_size}, chunk_stride: {self.chunk_partial_size}, total_chunk_num: {total_chunk_num}, len: {len(self.audio_cache)}")
+        end_idx = None
+        for i in range(total_chunk_num):
+            if auto_det_end:
+                is_end = i == total_chunk_num - 1
+            start_idx = i*self.chunk_partial_size
+            if auto_det_end:
+                end_idx = (i+1)*self.chunk_partial_size if i < total_chunk_num-1 else -1
+            else:
+                end_idx = (i+1)*self.chunk_partial_size if i < total_chunk_num else -1
+            # print(f"cut part: {start_idx}:{end_idx}, is_end: {is_end}, i: {i}, total_chunk_num: {total_chunk_num}")
+            # t_stamp = time.time()
+            
+            speech_chunk = self.audio_cache[start_idx:end_idx]
+
+            # TODO: exceptions processes
+            try:
+                res = self.asr_model.generate(input=speech_chunk, cache=self.asr_cache, is_final=is_end, chunk_size=self.chunk_size, encoder_chunk_look_back=self.encoder_chunk_look_back, decoder_chunk_look_back=self.decoder_chunk_look_back)
+            except ValueError as e:
+                print(f"ValueError: {e}")
+                continue
+            if self.use_punct:
+                text_dict['text'].append(self.puctuation_model.process(self.text_postprecess(res[0], data_id='text'), cache=text_dict))
+            else:
+                text_dict['text'].append(self.text_postprecess(res[0], data_id='text'))
+
+
+            # print(f"each chunk time: {time.time()-t_stamp}")
+            
+        if is_end:
+            self.audio_cache = None
+            self.asr_cache = {}
+        else:
+            if end_idx:
+                self.audio_cache = self.audio_cache[end_idx:] # cut the processed part from audio_cache
+        text_dict['is_end'] = is_end
+
+        if self.use_punct and is_end:
+            text_dict['text'].append(self.puctuation_model.process('#', cache=text_dict).replace('#', ''))
+        
+        # print(f"text_dict: {text_dict}")
+        return text_dict
\ No newline at end of file
diff --git a/takway/stt/punctuation_utils.py b/takway/stt/punctuation_utils.py
new file mode 100644
index 0000000..9e038e0
--- /dev/null
+++ b/takway/stt/punctuation_utils.py
@@ -0,0 +1,119 @@
+from funasr import AutoModel
+from modelscope.pipelines import pipeline
+from modelscope.utils.constant import Tasks
+
+PUNCTUATION_MARK = [",", ".", "?", "!", "，", "。", "？", "！"]
+"""
+FUNASR
+    模型大小: 1G
+    效果: 较好
+    输入类型: 仅支持字符串不支持list, 输入list会将list视为彼此独立的字符串处理
+"""
+FUNASR = {
+    "model_type": "funasr", 
+    "model_path": "ct-punc",
+    "model_revision": "v2.0.4"
+}
+"""
+CTTRANSFORMER
+    模型大小: 275M
+    效果：较差
+    输入类型: 支持字符串与list, 同时支持输入cache
+"""
+CTTRANSFORMER = {
+    "model_type": "ct-transformer",
+    "model_path": "iic/punc_ct-transformer_zh-cn-common-vad_realtime-vocab272727",
+    "model_revision": "v2.0.4"
+}
+
+class Punctuation:
+    def __init__(self,
+                 model_type="funasr", # funasr | ct-transformer
+                 model_path="ct-punc",
+                 device="cuda",
+                 model_revision="v2.0.4",
+                 **kwargs):
+        
+        self.model_type=model_type
+        self.initialize(model_type, model_path, device, model_revision, **kwargs)
+
+    def initialize(self,
+                   model_type,
+                   model_path,
+                   device,
+                   model_revision,
+                   **kwargs):
+        if model_type == 'funasr':
+            self.punc_model = AutoModel(model=model_path, device=device, model_revision=model_revision, **kwargs)
+        elif model_type == 'ct-transformer':
+            self.punc_model = pipeline(task=Tasks.punctuation, model=model_path, model_revision=model_revision, **kwargs)
+        else:
+            raise NotImplementedError(f"unsupported model type [{model_type}]. only [funasr|ct-transformer] expected.")
+
+    def check_text_type(self,
+                        text_data):
+        # funasr只支持单个str输入，不支持list输入，此处将list转化为字符串
+        if self.model_type == 'funasr':
+            if isinstance(text_data, str):
+                pass
+            elif isinstance(text_data, list):
+                text_data = ''.join(text_data)
+            else:
+                raise TypeError(f"text must be str or list, but got {type(list)}")
+        # ct-transformer支持list输入
+        # TODO 验证拆分字符串能否提高效率
+        elif self.model_type == 'ct-transformer':
+            if isinstance(text_data, str):
+                text_data = [text_data]
+            elif isinstance(text_data, list):
+                pass
+            else:
+                raise TypeError(f"text must be str or list, but got {type(list)}")
+        else:
+            pass
+        return text_data
+
+    def generate_cache(self, cache):
+        new_cache = {'pre_text': ""}
+        for text in cache['text']:
+            if text != '':
+                new_cache['pre_text'] = new_cache['pre_text']+text
+        return new_cache
+
+    def process(self,
+                text,
+                append_period=False,
+                cache={}):
+        if text == '':
+            return ''
+        text = self.check_text_type(text)
+        if self.model_type == 'funasr':
+            result = self.punc_model.generate(text)
+        elif self.model_type == 'ct-transformer':
+            if cache != {}:
+                cache = self.generate_cache(cache)
+            result = self.punc_model(text, cache=cache)
+        punced_text = ''
+        for res in result:
+            punced_text += res['text']
+        # 如果最后没有标点符号，手动加上。
+        if append_period and not punced_text[-1] in PUNCTUATION_MARK:
+            punced_text += "。"
+        return punced_text
+
+if __name__ == "__main__":
+    inputs = "把字符串拆分为list只|适用于ct-transformer模型|在数据处理部分|已经把list转为单个字符串"
+    """
+    把字符串拆分为list只适用于ct-transformer模型,
+    在数据处理部分,已经把list转为单个字符串
+    """
+    vads = inputs.split("|")
+    device = "cuda"
+    CTTRANSFORMER.update({"device": device})
+    puct_model = Punctuation(**CTTRANSFORMER)
+    result = puct_model.process(vads)
+    print(result)
+    # FUNASR.update({"device":"cuda"})
+    # puct_model = Punctuation(**FUNASR)   
+    # result = puct_model.process(vads)
+    # print(result)
\ No newline at end of file
diff --git a/takway/stt/speaker_ver_utils.py b/takway/stt/speaker_ver_utils.py
new file mode 100644
index 0000000..838393f
--- /dev/null
+++ b/takway/stt/speaker_ver_utils.py
@@ -0,0 +1,86 @@
+from modelscope.pipelines import pipeline
+import numpy as np
+import os
+import pdb
+ERES2NETV2 = {
+    "task": 'speaker-verification',
+    "model_name": 'damo/speech_eres2netv2_sv_zh-cn_16k-common',
+    "model_revision": 'v1.0.1',
+    "save_embeddings": False
+}
+
+# 保存 embedding 的路径
+DEFALUT_SAVE_PATH = r"D:\python\irving\takway_base-main\examples"
+
+class speaker_verfication:
+    def __init__(self, 
+                task='speaker-verification',
+                model_name='damo/speech_eres2netv2_sv_zh-cn_16k-common',
+                model_revision='v1.0.1',
+                device="cuda",
+                save_embeddings=False):
+        self.pipeline = pipeline(
+                task=task,
+                model=model_name,
+                model_revision=model_revision,
+                device=device)
+        self.save_embeddings = save_embeddings
+
+    def wav2embeddings(self, speaker_1_wav, save_path=None):
+        result = self.pipeline([speaker_1_wav], output_emb=True)
+        speaker_1_emb = result['embs'][0]
+        if save_path is not None:
+            np.save(save_path, speaker_1_emb)
+        return speaker_1_emb
+
+    def _verifaction(self, speaker_1_wav, speaker_2_wav, threshold, save_path):
+        if not self.save_embeddings:
+            result = self.pipeline([speaker_1_wav, speaker_2_wav], thr=threshold)
+            return result["text"]
+        else:
+            result = self.pipeline([speaker_1_wav, speaker_2_wav], thr=threshold, output_emb=True)
+            speaker1_emb = result["embs"][0]
+            speaker2_emb = result["embs"][1]
+            np.save(os.path.join(save_path, "speaker_1.npy"), speaker1_emb)
+            return result['outputs']["text"]
+        
+    def _verifaction_from_embedding(self, base_emb, speaker_2_wav, threshold):
+        base_emb = np.load(base_emb)
+        result = self.pipeline([speaker_2_wav], output_emb=True)
+        speaker2_emb = result["embs"][0]
+        similarity = np.dot(base_emb, speaker2_emb) / (np.linalg.norm(base_emb) * np.linalg.norm(speaker2_emb))
+        if similarity > threshold:
+            return "yes"
+        else:
+            return "no"
+    
+    def verfication(self, 
+                    base_emb=None, 
+                    speaker_1_wav=None, 
+                    speaker_2_wav=None, 
+                    threshold=0.333, 
+                    save_path=None):
+        if base_emb is not None and speaker_1_wav is not None:
+            raise ValueError("Only need one of them, base_emb or speaker_1_wav")
+        if base_emb is not None and speaker_2_wav is not None:
+            return self._verifaction_from_embedding(base_emb, speaker_2_wav, threshold)
+        elif speaker_1_wav is not None and speaker_2_wav is not None:
+            return self._verifaction(speaker_1_wav, speaker_2_wav, threshold, save_path)
+        else:
+            raise NotImplementedError
+        
+if __name__ == '__main__':
+    verifier = speaker_verfication(**ERES2NETV2)
+
+    verifier = speaker_verfication(save_embeddings=False)
+    result = verifier.verfication(base_emb=None, speaker_1_wav=r"C:\Users\bing\Downloads\speaker1_a_cn_16k.wav", 
+                                  speaker_2_wav=r"C:\Users\bing\Downloads\speaker2_a_cn_16k.wav",
+                                  threshold=0.333,
+                                  save_path=r"D:\python\irving\takway_base-main\savePath"
+                                  )
+    print("---")
+    print(result)
+    print(verifier.verfication(r"D:\python\irving\takway_base-main\savePath\speaker_1.npy",
+                               speaker_2_wav=r"C:\Users\bing\Downloads\speaker1_b_cn_16k.wav",
+                               threshold=0.333,
+                               ))
\ No newline at end of file
diff --git a/takway/stt/vosk_utils.py b/takway/stt/vosk_utils.py
new file mode 100644
index 0000000..b67cfa5
--- /dev/null
+++ b/takway/stt/vosk_utils.py
@@ -0,0 +1,120 @@
+#!/usr/bin/env python3
+# -*- coding: utf-8 -*-
+# ####################################################### #
+# VOSKAutoSpeechRecognizer
+# ####################################################### #
+import json
+import wave
+import io
+import os
+from vosk import Model, KaldiRecognizer, SetLogLevel
+from .base_stt import STTBase
+from ..common_utils import decode_str2bytes
+
+class VOSKAutoSpeechRecognizer(STTBase):
+    def __init__(self, model_path="vosk-model-small-cn-0.22", RATE=16000, cfg_path=None, efficent_mode=True, debug=False):
+        super().__init__(self, model_path=model_path, RATE=RATE, cfg_path=cfg_path, debug=debug)
+        self.asr_model = AutoModel(model="paraformer-zh-streaming")
+        
+        self.apply_asr_config(self.asr_cfg)
+    
+    def recognize_keywords(self, audio_data, partial_size=None, queue=None):
+        """recognize keywords in audio data"""
+        audio_data = self.check_audio_type(audio_data)
+        if partial_size is None:
+            rec_result = self.recognize(audio_data, queue)
+            rec_text = self.result_postprecess(rec_result)
+        else:
+            rec_result = self.partial_recognize(audio_data, partial_size, queue)
+            rec_text = self.result_postprecess(rec_result, 'partial')
+        print(f"rec_text: {rec_text}")
+        if rec_text != '':
+            print(f"rec_text: {rec_text}")
+        if any(keyword in rec_text for keyword in self.keywords):
+            print("Keyword detected.")
+            return True, rec_text
+        else:
+            return False, None
+
+    def recognize(self, audio_data, queue=None):
+        """recognize audio data to text"""
+        audio_data = self.check_audio_type(audio_data)
+        self.asr.AcceptWaveform(audio_data)
+        result = json.loads(self.asr.FinalResult())
+        # TODO: put result to queue
+        return result
+
+    def partial_recognize(self, audio_data, partial_size=1024, queue=None):
+        """recognize partial result"""
+        audio_data = self.check_audio_type(audio_data)
+        text_dict = dict(
+            text=[],
+            partial=[],
+            final=[],
+            is_end=False)
+        # 逐个分割音频数据进行识别
+        for i in range(0, len(audio_data), partial_size):
+            # print(f"partial data: {i} - {i+partial_size}")
+            data = audio_data[i:i+partial_size]
+            if len(data) == 0:
+                break
+            if self.asr.AcceptWaveform(data):
+                result = json.loads(self.asr.Result())
+                if result['text'] != '':
+                    text_dict['text'].append(result['text'])
+                    if queue is not None:
+                        queue.put(('stt_info', text_dict))
+                    # print(f"text result: {result}")
+            else:
+                result = json.loads(self.asr.PartialResult())
+                if result['partial'] != '':
+                    # text_dict['partial'].append(result['partial'])
+                    text_dict['partial'] = [result['partial']]
+                    if queue is not None:
+                        queue.put(('stt_info', text_dict))
+                    # print(f"partial result: {result}")
+        
+        # final recognize
+        final_result = json.loads(self.asr.FinalResult())
+        if final_result['text'] != '':
+            text_dict['final'].append(final_result['text'])
+            text_dict['text'].append(final_result['text'])
+            
+        text_dict['is_end'] = True
+        
+        print(f"final dict: {text_dict}")
+        if queue is not None:
+            queue.put(('stt_info', text_dict))
+        return text_dict
+        
+
+if __name__ == "__main__":
+    '''
+    wav_file_path = "recording.wav"
+
+    # You can set log level to -1 to disable debug messages
+    SetLogLevel(0)
+
+    model = Model(model_path="vosk-model-small-cn-0.22")
+
+    # 调用函数进行录音
+    # record_audio(wav_file_path)
+    data = record_audio()
+
+    # 调用函数进行音频转写
+    result = audio_to_text(data, model)
+
+    print("-------------")
+    print(result)
+    '''
+    from takway.audio_utils import Recorder
+    rec = Recorder()
+    
+    return_type = 'bytes'
+    data = rec.record(return_type)
+    print(type(data))
+        
+    asr = AutoSpeechRecognizer()
+    # asr.recognize(data)
+    asr.add_keyword("你好")
+    asr.recognize_keywords(data)
\ No newline at end of file