add frontend demo

Files changed (9) hide show

.gitignore +1 -1
api_model.py +23 -0
config.py +1 -1
frontend/index.html +169 -0
pyproject.toml +1 -0
transcribe/transcription.py +14 -12
transcribe/translator.py +1 -1
transcribe/whisper_llm_serve.py +83 -21
uv.lock +103 -0

.gitignore CHANGED Viewed

@@ -3,7 +3,7 @@ __pycache__/
 *$py.class
 # C extensions
 *.so
 # Distribution / packaging
 .Python
 build/

 *$py.class
 # C extensions
 *.so
+.DS_Store
 # Distribution / packaging
 .Python
 build/

api_model.py ADDED Viewed

	@@ -0,0 +1,23 @@

+from pydantic import BaseModel, Field
+from uuid import UUID
+class TransResult(BaseModel):
+    # trans_pattern: str
+    seg_id: int
+    context: str
+    from_: str = Field(alias="from")
+    to: str
+    tran_content: str = Field(alias="tranContent")
+    partial: bool = True
+    class Config:
+        populate_by_name = True
+class Message(BaseModel):
+    result: TransResult = {}
+    # action: str = "recognition"
+    error_code: int = 0
+    request_id: UUID

config.py CHANGED Viewed

@@ -9,7 +9,7 @@ PAUSE_END_MARKERS = [',', '，', '、']
 # whisper推理参数
 WHISPER_PROMPT_ZH = "以下是简体中文普通话的句子。"
-MAX_LENTH_ZH = 3
 WHISPER_PROMPT_EN = "The following is an English sentence."
 MAX_LENGTH_EN= 1

 # whisper推理参数
 WHISPER_PROMPT_ZH = "以下是简体中文普通话的句子。"
+MAX_LENTH_ZH = 4
 WHISPER_PROMPT_EN = "The following is an English sentence."
 MAX_LENGTH_EN= 1

frontend/index.html ADDED Viewed

	@@ -0,0 +1,169 @@

+<!DOCTYPE html>
+<html lang="zh">
+<head>
+  <meta charset="UTF-8" />
+  <title>试试翻译</title>
+  <style>
+    body {
+      background-color: #f9f9fc;
+      font-family: -apple-system, BlinkMacSystemFont, "Segoe UI", Roboto, "Helvetica Neue", sans-serif;
+      margin: 0;
+      padding: 2rem;
+    }
+    .translation-box {
+      background: #f2f2f8;
+      border-radius: 12px;
+      padding: 1.5rem;
+      max-width: 800px;
+      margin: 0 auto;
+      min-height: 200px;
+    }
+    .entry {
+      margin-bottom: 1.5rem;
+    }
+    .timestamp {
+      font-size: 0.75rem;
+      color: #999;
+    }
+    .original {
+      font-size: 1rem;
+      color: #333;
+    }
+    .translation {
+      font-size: 1rem;
+      font-weight: bold;
+      color: #000;
+    }
+    .footer {
+      display: flex;
+      justify-content: space-between;
+      align-items: center;
+      margin-top: 2rem;
+    }
+    .lang-select {
+      background: white;
+      border-radius: 9999px;
+      padding: 0.4rem 1rem;
+      border: none;
+      font-size: 1rem;
+      box-shadow: 0 0 0 1px #ddd;
+    }
+    .record-button {
+      background-color: #1e40af;
+      color: white;
+      border: none;
+      padding: 0.6rem 1.2rem;
+      border-radius: 9999px;
+      font-size: 1rem;
+      cursor: pointer;
+    }
+  </style>
+</head>
+<body>
+  <div class="translation-box" id="translationBox">
+    <!-- 实时内容将插入这里 -->
+  </div>
+  <div class="footer">
+    <select class="lang-select">
+      <option>中文 » 英语</option>
+    </select>
+    <button class="record-button" onclick="startRecording()">🎤 录音</button>
+  </div>
+  <script>
+    let ws;
+    let mediaRecorder;
+    function formatTimestamp(ms) {
+      const sec = ms / 1000;
+      const min = Math.floor(sec / 60);
+      const s = (sec % 60).toFixed(1);
+      return `${String(min).padStart(2, '0')}:${s.padStart(4, '0')}`;
+    }
+    let lastSegId = null;  // 用于存储上一个 seg_id
+function addTranslation(result) {
+  const box = document.getElementById('translationBox');
+  // 创建一个新的 div 来显示翻译
+  const entry = document.createElement('div');
+  entry.className = 'entry';
+  console.log(result);
+  const start = formatTimestamp(result.bg);
+  const end = formatTimestamp(result.ed);
+  // 判断是否是新的一行
+  if (result.seg_id === lastSegId) {
+    // 如果 seg_id 相同，更新该行内容
+    const existingEntry = box.querySelector(`.entry[data-seg-id="${result.seg_id}"]`);
+    if (existingEntry) {
+      const translationDiv = existingEntry.querySelector('.translation');
+      translationDiv.innerHTML = result.tranContent;
+    }
+  } else {
+    // 如果 seg_id 不同，表示是新的行，添加新行
+    entry.setAttribute('data-seg-id', result.seg_id);  // 设置 seg_id
+    entry.innerHTML = `
+      <div class="original">${result.context}</div>
+      <div class="translation">${result.tranContent}</div>
+    `;
+    box.appendChild(entry);
+  }
+  // 更新 lastSegId 以便下一次判断
+  lastSegId = result.seg_id;
+}
+    async function startRecording() {
+      const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
+      const audioContext = new AudioContext({ sampleRate: 16000 });
+      const source = audioContext.createMediaStreamSource(stream);
+      const processor = audioContext.createScriptProcessor(4096, 1, 1);
+      const wsUrl = "ws://localhost:9090?from=zh&to=en";
+      ws = new WebSocket(wsUrl);
+      ws.binaryType = "arraybuffer";
+      ws.onopen = () => {
+        console.log("WebSocket opened");
+        source.connect(processor);
+        processor.connect(audioContext.destination);
+        processor.onaudioprocess = (e) => {
+          const input = e.inputBuffer.getChannelData(0);
+          const buffer = new Int16Array(input.length);
+          for (let i = 0; i < input.length; i++) {
+            buffer[i] = Math.max(-1, Math.min(1, input[i])) * 0x7FFF;
+          }
+          ws.send(buffer);
+        };
+      };
+      ws.onmessage = (event) => {
+        try {
+          const msg = JSON.parse(event.data);
+          if (msg.result) {
+            addTranslation(msg.result);
+          }
+        } catch (e) {
+          console.error("Parse error:", e);
+        }
+      };
+      ws.onerror = (e) => console.error("WebSocket error:", e);
+      ws.onclose = () => {
+        console.log("WebSocket closed");
+        processor.disconnect();
+        source.disconnect();
+      };
+    }
+  </script>
+</body>
+</html>

pyproject.toml CHANGED Viewed

@@ -11,6 +11,7 @@ dependencies = [
     "numpy>=2.1.3",
     "onnxruntime>=1.21.0",
     "pyaudio>=0.2.14",
     "pywhispercpp>=1.3.0",
     "setuptools>=78.1.0",
     "soundfile>=0.13.1",

     "numpy>=2.1.3",
     "onnxruntime>=1.21.0",
     "pyaudio>=0.2.14",
+    "pydantic>=2.11.2",
     "pywhispercpp>=1.3.0",
     "setuptools>=78.1.0",
     "soundfile>=0.13.1",

transcribe/transcription.py CHANGED Viewed

@@ -13,7 +13,7 @@ from .vad import VoiceActivityDetector
 from urllib.parse import urlparse, parse_qsl
 from websockets.exceptions import ConnectionClosed
 from websockets.sync.server import serve
 logging.basicConfig(level=logging.INFO)
@@ -174,14 +174,19 @@ class TranscriptionServer:
         frame_data = websocket.recv()
         if frame_data == b"END_OF_AUDIO":
             return False
-        return np.frombuffer(frame_data, dtype=np.float32)
     def handle_new_connection(self, websocket):
         try:
             logging.info("New client connected")
-            options = websocket.recv()
-            options = json.loads(options)
             if self.client_manager is None:
                 max_clients = options.get('max_clients', 4)
                 max_connection_time = options.get('max_connection_time', 600)
@@ -195,7 +200,9 @@ class TranscriptionServer:
                 self.vad_detector = VoiceActivityDetector(frame_rate=self.RATE)
             self.initialize_client(websocket, options)
             return True
         except json.JSONDecodeError:
             logging.error("Failed to decode JSON from client")
@@ -240,12 +247,7 @@ class TranscriptionServer:
         self.backend = backend
         if not self.handle_new_connection(websocket):
             return
-        query_parameters_dict = dict(parse_qsl(urlparse(websocket.request.path).query))
-        from_lang, to_lang = query_parameters_dict.get('from'), query_parameters_dict.get('to')
-        if from_lang and to_lang:
-            self.set_lang(websocket, from_lang, to_lang)
-            logging.info(f"Source lange: {from_lang}  -> Dst lange: {to_lang}")
         try:
             while not self.client_manager.is_client_timeout(websocket):

 from urllib.parse import urlparse, parse_qsl
 from websockets.exceptions import ConnectionClosed
 from websockets.sync.server import serve
+from uuid import uuid1
 logging.basicConfig(level=logging.INFO)
         frame_data = websocket.recv()
         if frame_data == b"END_OF_AUDIO":
             return False
+        return np.frombuffer(frame_data, dtype=np.int16).astype(np.float32) / 32768.0
     def handle_new_connection(self, websocket):
+        query_parameters_dict = dict(parse_qsl(urlparse(websocket.request.path).query))
+        from_lang, to_lang = query_parameters_dict.get('from'), query_parameters_dict.get('to')
         try:
             logging.info("New client connected")
+            # options = websocket.recv()
+            # options = json.loads(options)
+            options = {"language": from_lang, "uid": str(uuid1())}
             if self.client_manager is None:
                 max_clients = options.get('max_clients', 4)
                 max_connection_time = options.get('max_connection_time', 600)
                 self.vad_detector = VoiceActivityDetector(frame_rate=self.RATE)
             self.initialize_client(websocket, options)
+            if from_lang and to_lang:
+                self.set_lang(websocket, from_lang, to_lang)
+                logging.info(f"Source lange: {from_lang}  -> Dst lange: {to_lang}")
             return True
         except json.JSONDecodeError:
             logging.error("Failed to decode JSON from client")
         self.backend = backend
         if not self.handle_new_connection(websocket):
             return
         try:
             while not self.client_manager.is_client_timeout(websocket):

transcribe/translator.py CHANGED Viewed

@@ -28,7 +28,7 @@ class QwenTranslator:
         message = self.to_message(prompt, src_lang, dst_lang)
         start_time = time.monotonic()
         output = self.llm.create_chat_completion(messages=message, temperature=0.9)
-        logger.info(f"LLM inference time: {time.monotonic() - start_time:.2f}s.")
         return output['choices'][0]['message']['content']
     def __call__(self, prompt,*args, **kwargs):

         message = self.to_message(prompt, src_lang, dst_lang)
         start_time = time.monotonic()
         output = self.llm.create_chat_completion(messages=message, temperature=0.9)
+        logger.info(f"Translate time: {time.monotonic() - start_time:.2f}s.")
         return output['choices'][0]['message']['content']
     def __call__(self, prompt,*args, **kwargs):

transcribe/whisper_llm_serve.py CHANGED Viewed

@@ -1,7 +1,7 @@
 import soundfile
-from concurrent.futures import ProcessPoolExecutor as Pool
 import multiprocessing as mp
 import numpy as np
 from logging import getLogger
@@ -14,10 +14,17 @@ import threading
 from functools import partial
 from .server import ServeClientBase
 from .translator import QwenTranslator
 from pywhispercpp.model import Model
 logger = getLogger(__name__)
 class TripleTextBuffer:
     def __init__(self, size=2):
@@ -108,6 +115,7 @@ class SegmentManager:
 class PywhisperInference:
     whisper_model = None
     llm_model = None
     @classmethod
     def initializer(cls, event:mp.Event, warmup=True):
@@ -124,6 +132,7 @@ class PywhisperInference:
         # init llamacpp
         cls.llm_model = QwenTranslator(config.LLM_MODEL_PATH, config.LLM_SYS_PROMPT)
         event.set()
     @classmethod
@@ -155,6 +164,12 @@ class PywhisperInference:
     @classmethod
     def translate(cls, context: str, src_lang, dst_lang):
         return cls.llm_model.translate(context, src_lang, dst_lang)
 class PyWhiperCppServe(ServeClientBase):
@@ -171,16 +186,21 @@ class PyWhiperCppServe(ServeClientBase):
         self.lock = threading.Lock()
         self.frames_np = None
         self.sample_rate = 16000
         self._ready_state = mp.Event()
-        self._pool = Pool(
             max_workers=1, initializer=partial(PywhisperInference.initializer, event=self._ready_state))
         logger.info('Create a process to process audio.')
         self.trans_thread = threading.Thread(target=self.speech_to_text)
         self.trans_thread.daemon = True
         self.trans_thread.start()
-        self.send_ready_state()
     def send_ready_state(self):
         while not self._ready_state:
@@ -196,13 +216,29 @@ class PyWhiperCppServe(ServeClientBase):
         self.language = src_lang
         self.dst_lang = dst_lang
     def add_frames(self, frame_np):
         with self.lock:
             if self.frames_np is None:
                 self.frames_np = frame_np.copy()
             else:
                 self.frames_np = np.append(self.frames_np,frame_np)
     def update_audio_buffer(self, last_offset):
         with self.lock:
@@ -273,6 +309,7 @@ class PyWhiperCppServe(ServeClientBase):
         return None, left_watch_string, right_watch_string, is_end_sentence
     def speech_to_text(self):
         while True:
             if self.exit:
                 logger.info("Exiting speech to text thread")
@@ -284,13 +321,17 @@ class PyWhiperCppServe(ServeClientBase):
                 continue
             audio_buffer = self.get_audio_chunk_for_processing()
-            # logger.info(f"[pywhispercpp:] Processing audio with duration: {len(audio_buffer)}")
             # segments = self.transcribe_audio(audio_buffer)
             try:
                 logger.info(f"Processing audio with duration: {len(audio_buffer) / self.sample_rate:.2f}s")
                 segments = self.transcribe_audio(audio_buffer)
-                for item in self.handle_transcription_output(segments, audio_buffer):
-                    print(item)
             except KeyboardInterrupt:
                 break
             except Exception as e:
@@ -300,6 +341,8 @@ class PyWhiperCppServe(ServeClientBase):
     def handle_transcription_output(self, segments, audio_buffer):
         texts  = "".join(i.text for i in segments)
         self._segment_manager.handle(texts)
          # 分析句子
         last_cut_index, left_string, right_string, is_end_sentence = self.analysis_segments(segments, audio_buffer)
@@ -313,26 +356,45 @@ class PyWhiperCppServe(ServeClientBase):
         if is_end_sentence and last_cut_index:
             message = self._segment_manager.segment
             seg_id = self._segment_manager.get_seg_id() - 1
-            yield (seg_id, message, self.translate_text(message))
             if self._segment_manager.string.strip():
-                yield (seg_id + 1, self._segment_manager.string, self.translate_text(self._segment_manager.string))
         else:
             seg_id = self._segment_manager.get_seg_id()
             message = self._segment_manager.short_sentence + self._segment_manager.string
-            # print(self._segment_manager.__dict__)
-            # elapsed_time = time.time() - start_time
-            # formatted_time = f"{int(elapsed_time // 60):02}:{int(elapsed_time % 60):02}:{(elapsed_time % 1) * 1000:03.0f}"
-            yield (seg_id, message,self.translate_text(message))
-    def send_to_client(self, data_dict):
-        content = {
-            "uid": self.client_uid,
-            **data_dict
-        }
         try:
             self.websocket.send(
-                json.dumps(content)
             )
         except Exception as e:
             logger.error(f"Sending data to client: {e}")

 import soundfile
+from concurrent.futures import ProcessPoolExecutor as PPool
 import multiprocessing as mp
 import numpy as np
 from logging import getLogger
 from functools import partial
 from .server import ServeClientBase
 from .translator import QwenTranslator
+from .vad import VoiceActivityDetector
 from pywhispercpp.model import Model
+from queue import Queue
+from scipy.io.wavfile import write
+from api_model import TransResult, Message
 logger = getLogger(__name__)
+def save_to_wave(filename, data:np.ndarray, sample_rate=16000):
+    write(filename, sample_rate, data)
 class TripleTextBuffer:
     def __init__(self, size=2):
 class PywhisperInference:
     whisper_model = None
     llm_model = None
+    vad_model = None
     @classmethod
     def initializer(cls, event:mp.Event, warmup=True):
         # init llamacpp
         cls.llm_model = QwenTranslator(config.LLM_MODEL_PATH, config.LLM_SYS_PROMPT)
+        cls.vad_model = VoiceActivityDetector()
         event.set()
     @classmethod
     @classmethod
     def translate(cls, context: str, src_lang, dst_lang):
         return cls.llm_model.translate(context, src_lang, dst_lang)
+    @classmethod
+    def voice_detect(cls, audio_buffer):
+        audio_buffer = np.frombuffer(audio_buffer, dtype=np.float32)
+        return cls.vad_model(audio_buffer)
 class PyWhiperCppServe(ServeClientBase):
         self.lock = threading.Lock()
         self.frames_np = None
         self.sample_rate = 16000
+        # self._audio_queue = Queue()
+        # 进程初始化后再开始收音频
         self._ready_state = mp.Event()
+        self._pool = PPool(
             max_workers=1, initializer=partial(PywhisperInference.initializer, event=self._ready_state))
         logger.info('Create a process to process audio.')
+        self.send_ready_state()
+        # self.load_frame_thread = threading.Thread(target=self.load_frame_from_queue)
+        # self.load_frame_thread.daemon = True
+        # self.load_frame_thread.start()
         self.trans_thread = threading.Thread(target=self.speech_to_text)
         self.trans_thread.daemon = True
         self.trans_thread.start()
     def send_ready_state(self):
         while not self._ready_state:
         self.language = src_lang
         self.dst_lang = dst_lang
+    # def load_frame_from_queue(self):
+    #     while True:
+    #         frame_np = self._audio_queue.get()
+    #         fut = self._pool.submit(PywhisperInference.voice_detect, frame_np.tobytes())
+    #         output = fut.result()
+    #         logger.info(f"VAD: {output}")
+    #         if output == True:
+    #             with self.lock:
+    #                 if self.frames_np is None:
+    #                     self.frames_np = frame_np.copy()
+    #                 else:
+    #                     self.frames_np = np.append(self.frames_np,frame_np)
     def add_frames(self, frame_np):
+        # self._audio_queue.put(frame_np)
         with self.lock:
             if self.frames_np is None:
                 self.frames_np = frame_np.copy()
             else:
                 self.frames_np = np.append(self.frames_np,frame_np)
     def update_audio_buffer(self, last_offset):
         with self.lock:
         return None, left_watch_string, right_watch_string, is_end_sentence
     def speech_to_text(self):
+        # c = 0
         while True:
             if self.exit:
                 logger.info("Exiting speech to text thread")
                 continue
             audio_buffer = self.get_audio_chunk_for_processing()
+            logger.info(f"Processing audio with duration: {len(audio_buffer)}")
             # segments = self.transcribe_audio(audio_buffer)
+            # c+= 1
+            # name = f"dev-{c}.wav"
+            # save_to_wave(name, audio_buffer)
             try:
                 logger.info(f"Processing audio with duration: {len(audio_buffer) / self.sample_rate:.2f}s")
                 segments = self.transcribe_audio(audio_buffer)
+                for tran_result in self.handle_transcription_output(segments, audio_buffer):
+                    self.send_to_client(tran_result)
             except KeyboardInterrupt:
                 break
             except Exception as e:
     def handle_transcription_output(self, segments, audio_buffer):
         texts  = "".join(i.text for i in segments)
+        if not len(texts):
+            return
         self._segment_manager.handle(texts)
          # 分析句子
         last_cut_index, left_string, right_string, is_end_sentence = self.analysis_segments(segments, audio_buffer)
         if is_end_sentence and last_cut_index:
             message = self._segment_manager.segment
             seg_id = self._segment_manager.get_seg_id() - 1
+            logger.info(f"{seg_id}, {message}")
+            yield TransResult(
+                seg_id=seg_id,
+                context=message,
+                from_=self.language,
+                to=self.dst_lang,
+                tran_content="this is english sample",
+                partial=False
+            )
             if self._segment_manager.string.strip():
+                logger.info(f"{seg_id + 1}, {self._segment_manager.string}")
+                yield TransResult(
+                    seg_id=seg_id+1,
+                    context=self._segment_manager.string,
+                    from_=self.language,
+                    to=self.dst_lang,
+                    tran_content="this is english sample",
+            )
         else:
             seg_id = self._segment_manager.get_seg_id()
             message = self._segment_manager.short_sentence + self._segment_manager.string
+            logger.info(f"{seg_id}, {message}")
+            yield TransResult(
+                seg_id=seg_id,
+                context=message,
+                from_=self.language,
+                to=self.dst_lang,
+                tran_content="this is english sample",
+            )
+    def send_to_client(self, data:TransResult):
+        # content = {
+        #     "uid": self.client_uid,
+        #     **data_dict
+        # }
         try:
             self.websocket.send(
+               Message(result=data, request_id=self.client_uid).model_dump_json(by_alias=True)
             )
         except Exception as e:
             logger.error(f"Sending data to client: {e}")

uv.lock CHANGED Viewed

@@ -7,6 +7,15 @@ resolution-markers = [
     "python_full_version < '3.12'",
 ]
 [[package]]
 name = "audioop-lts"
 version = "0.2.1"
@@ -763,6 +772,86 @@ wheels = [
     { url = "https://files.pythonhosted.org/packages/13/a3/a812df4e2dd5696d1f351d58b8fe16a405b234ad2886a0dab9183fb78109/pycparser-2.22-py3-none-any.whl", hash = "sha256:c3702b6d3dd8c7abc1afa565d7e63d53a1d0bd86cdc24edd75470f4de499cfcc", size = 117552 },
 ]
 [[package]]
 name = "pyreadline3"
 version = "3.5.4"
@@ -1067,6 +1156,7 @@ dependencies = [
     { name = "numpy" },
     { name = "onnxruntime" },
     { name = "pyaudio" },
     { name = "pywhispercpp" },
     { name = "setuptools" },
     { name = "soundfile" },
@@ -1084,6 +1174,7 @@ requires-dist = [
     { name = "numpy", specifier = ">=2.1.3" },
     { name = "onnxruntime", specifier = ">=1.21.0" },
     { name = "pyaudio", specifier = ">=0.2.14" },
     { name = "pywhispercpp", specifier = ">=1.3.0" },
     { name = "setuptools", specifier = ">=78.1.0" },
     { name = "soundfile", specifier = ">=0.13.1" },
@@ -1112,6 +1203,18 @@ wheels = [
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