add diarization model list url

Vac update

LICENSE

@@ -1,10 +1,6 @@
 MIT License
 
 Copyright (c) 2025 Quentin Fuxa.  
-Based on:
-- The original work by ÚFAL. License: https://github.com/ufal/whisper_streaming/blob/main/LICENSE  
-- The work by Snakers4 (silero-vad). License: https://github.com/snakers4/silero-vad/blob/f6b1294cb27590fb2452899df98fb234dfef1134/LICENSE  
-- The work in Diart by juanmc2005. License: https://github.com/juanmc2005/diart/blob/main/LICENSE
 
 Permission is hereby granted, free of charge, to any person obtaining a copy  
 of this software and associated documentation files (the "Software"), to deal  
@@ -26,8 +22,7 @@ SOFTWARE.
 
 ---
 
-Third-party components included in this software:
+Based on:
-
+- **whisper_streaming** by ÚFAL – MIT License – https://github.com/ufal/whisper_streaming. The original work by ÚFAL. License: https://github.com/ufal/whisper_streaming/blob/main/LICENSE  
-- **whisper_streaming** by ÚFAL – MIT License – https://github.com/ufal/whisper_streaming  
+- **silero-vad** by Snakers4 – MIT License – https://github.com/snakers4/silero-vad. The work by Snakers4 (silero-vad). License: https://github.com/snakers4/silero-vad/blob/f6b1294cb27590fb2452899df98fb234dfef1134/LICENSE  
-- **silero-vad** by Snakers4 – MIT License – https://github.com/snakers4/silero-vad  
+- **Diart** by juanmc2005 – MIT License – https://github.com/juanmc2005/diart. The work in Diart by juanmc2005. License: https://github.com/juanmc2005/diart/blob/main/LICENSE
-- **Diart** by juanmc2005 – MIT License – https://github.com/juanmc2005/diart

README.md

@@ -9,8 +9,8 @@
 <p align="center">
   <a href="https://pypi.org/project/whisperlivekit/"><img alt="PyPI Version" src="https://img.shields.io/pypi/v/whisperlivekit?color=g"></a>
   <a href="https://pepy.tech/project/whisperlivekit"><img alt="PyPI Downloads" src="https://static.pepy.tech/personalized-badge/whisperlivekit?period=total&units=international_system&left_color=grey&right_color=brightgreen&left_text=downloads"></a>
-  <a href="https://pypi.org/project/whisperlivekit/"><img alt="Python Versions" src="https://img.shields.io/badge/python-3.9%20%7C%203.10%20%7C%203.11%20%7C%203.12-dark_green"></a>
+  <a href="https://pypi.org/project/whisperlivekit/"><img alt="Python Versions" src="https://img.shields.io/badge/python-3.9--3.13-dark_green"></a>
-  <a href="https://github.com/QuentinFuxa/WhisperLiveKit/blob/main/LICENSE"><img alt="License" src="https://img.shields.io/github/license/QuentinFuxa/WhisperLiveKit?color=blue"></a>
+  <a href="https://github.com/QuentinFuxa/WhisperLiveKit/blob/main/LICENSE"><img alt="License" src="https://img.shields.io/badge/License-MIT-dark_green"></a>
 </p>
 
 ## 🚀 Overview
@@ -32,6 +32,7 @@ WhisperLiveKit consists of three main components:
 - **👥 Speaker Diarization** - Identify different speakers in real-time using [Diart](https://github.com/juanmc2005/diart)
 - **🔒 Fully Local** - All processing happens on your machine - no data sent to external servers
 - **📱 Multi-User Support** - Handle multiple users simultaneously with a single backend/server
+- **📝 Punctuation-Based Speaker Splitting [BETA] ** - Align speaker changes with natural sentence boundaries for more readable transcripts
  
 ### ⚙️ Core differences from [Whisper Streaming](https://github.com/ufal/whisper_streaming)
 
@@ -112,9 +113,6 @@ pip install whisperlivekit[whisper]              # Original Whisper
 pip install whisperlivekit[whisper-timestamped]  # Improved timestamps
 pip install whisperlivekit[mlx-whisper]          # Apple Silicon optimization
 pip install whisperlivekit[openai]               # OpenAI API
-
-# System audio capture (Windows only)
-pip install whisperlivekit[pyaudiowpatch]        # Use PyAudioWPatch for system audio loopback
 ```
 
 ### 🎹 Pyannote Models Setup
@@ -142,58 +140,82 @@ whisperlivekit-server --model tiny.en
 
 # Advanced configuration with diarization
 whisperlivekit-server --host 0.0.0.0 --port 8000 --model medium --diarization --language auto
-
-# Using PyAudioWPatch for system audio input (Windows only)
-whisperlivekit-server --model tiny.en --audio-input pyaudiowpatch
 ```
 
 ### Python API Integration (Backend)
+Check [basic_server.py](https://github.com/QuentinFuxa/WhisperLiveKit/blob/main/whisperlivekit/basic_server.py) for a complete example.
 
 ```python
-from whisperlivekit import WhisperLiveKit
+from whisperlivekit import TranscriptionEngine, AudioProcessor, get_web_interface_html, parse_args
-from whisperlivekit.audio_processor import AudioProcessor
+from fastapi import FastAPI, WebSocket, WebSocketDisconnect
-from fastapi import FastAPI, WebSocket
-import asyncio
 from fastapi.responses import HTMLResponse
+from contextlib import asynccontextmanager
+import asyncio
 
-# Initialize components
+# Global variable for the transcription engine
-app = FastAPI()
+transcription_engine = None
-kit = WhisperLiveKit(model="medium", diarization=True)
+
+@asynccontextmanager
+async def lifespan(app: FastAPI):
+    global transcription_engine
+    # Example: Initialize with specific parameters directly
+    # You can also load from command-line arguments using parse_args()
+    # args = parse_args()
+    # transcription_engine = TranscriptionEngine(**vars(args))
+    transcription_engine = TranscriptionEngine(model="medium", diarization=True, lan="en")
+    yield
+
+app = FastAPI(lifespan=lifespan)
 
 # Serve the web interface
 @app.get("/")
 async def get():
-    return HTMLResponse(kit.web_interface())  # Use the built-in web interface
+    return HTMLResponse(get_web_interface_html())
 
 # Process WebSocket connections
-async def handle_websocket_results(websocket, results_generator):
+async def handle_websocket_results(websocket: WebSocket, results_generator):
-    async for response in results_generator:
+    try:
-        await websocket.send_json(response)
+        async for response in results_generator:
+            await websocket.send_json(response)
+        await websocket.send_json({"type": "ready_to_stop"})
+    except WebSocketDisconnect:
+        print("WebSocket disconnected during results handling.")
 
 @app.websocket("/asr")
 async def websocket_endpoint(websocket: WebSocket):
-    audio_processor = AudioProcessor()
+    global transcription_engine
-    await websocket.accept()
-    results_generator = await audio_processor.create_tasks()
-    websocket_task = asyncio.create_task(
-        handle_websocket_results(websocket, results_generator)
-    )
 
+    # Create a new AudioProcessor for each connection, passing the shared engine
+    audio_processor = AudioProcessor(transcription_engine=transcription_engine)    
+    results_generator = await audio_processor.create_tasks()
+    send_results_to_client = handle_websocket_results(websocket, results_generator)
+    results_task = asyncio.create_task(send_results_to_client)
+    await websocket.accept()
     try:
         while True:
             message = await websocket.receive_bytes()
-            await audio_processor.process_audio(message)
+            await audio_processor.process_audio(message)        
+    except WebSocketDisconnect:
+        print(f"Client disconnected: {websocket.client}")
     except Exception as e:
-        print(f"WebSocket error: {e}")
+        await websocket.close(code=1011, reason=f"Server error: {e}")
-        websocket_task.cancel()
+    finally:
+        results_task.cancel()
+        try:
+            await results_task
+        except asyncio.CancelledError:
+            logger.info("Results task successfully cancelled.")
 ```
 
 ### Frontend Implementation
 
-The package includes a simple HTML/JavaScript implementation that you can adapt for your project. You can get in in [whisperlivekit/web/live_transcription.html](https://github.com/QuentinFuxa/WhisperLiveKit/blob/main/whisperlivekit/web/live_transcription.html), or using :
+The package includes a simple HTML/JavaScript implementation that you can adapt for your project. You can find it in `whisperlivekit/web/live_transcription.html`, or load its content using the `get_web_interface_html()` function from `whisperlivekit`:
 
 ```python
-kit.web_interface()
+from whisperlivekit import get_web_interface_html
+
+# ... later in your code where you need the HTML string ...
+html_content = get_web_interface_html()
 ```
 
 ## ⚙️ Configuration Reference
@@ -209,15 +231,17 @@ WhisperLiveKit offers extensive configuration options:
 | `--task` | `transcribe` or `translate` | `transcribe` |
 | `--backend` | Processing backend | `faster-whisper` |
 | `--diarization` | Enable speaker identification | `False` |
+| `--punctuation-split` | Use punctuation to improve speaker boundaries | `True` |
 | `--confidence-validation` | Use confidence scores for faster validation | `False` |
 | `--min-chunk-size` | Minimum audio chunk size (seconds) | `1.0` |
 | `--vac` | Use Voice Activity Controller | `False` |
 | `--no-vad` | Disable Voice Activity Detection | `False` |
 | `--buffer_trimming` | Buffer trimming strategy (`sentence` or `segment`) | `segment` |
 | `--warmup-file` | Audio file path for model warmup | `jfk.wav` |
-| `--audio-input` | Source of audio (`websocket` or `pyaudiowpatch`) | `websocket` |
 | `--ssl-certfile` | Path to the SSL certificate file (for HTTPS support) | `None` |
 | `--ssl-keyfile` | Path to the SSL private key file (for HTTPS support) | `None` |
+| `--segmentation-model` | Hugging Face model ID for pyannote.audio segmentation model. [Available models](https://github.com/juanmc2005/diart/tree/main?tab=readme-ov-file#pre-trained-models) | `pyannote/segmentation-3.0` |
+| `--embedding-model` | Hugging Face model ID for pyannote.audio embedding model. [Available models](https://github.com/juanmc2005/diart/tree/main?tab=readme-ov-file#pre-trained-models) | `speechbrain/spkrec-ecapa-voxceleb` |
 
 ## 🔧 How It Works
 
@@ -225,16 +249,12 @@ WhisperLiveKit offers extensive configuration options:
   <img src="https://raw.githubusercontent.com/QuentinFuxa/WhisperLiveKit/refs/heads/main/demo.png" alt="WhisperLiveKit in Action" width="500">
 </p>
 
-1. **Audio Input**:
+1. **Audio Capture**: Browser's MediaRecorder API captures audio in webm/opus format
-   - **WebSocket (Default)**: Browser's MediaRecorder API captures audio (webm/opus), streams via WebSocket.
+2. **Streaming**: Audio chunks are sent to the server via WebSocket
-   - **PyAudioWPatch (Windows Only)**: Captures system audio output directly using WASAPI loopback. Requires `--audio-input pyaudiowpatch`.
+3. **Processing**: Server decodes audio with FFmpeg and streams into Whisper for transcription
-2. **Processing**:
+4. **Real-time Output**: 
-   - **WebSocket**: Server decodes webm/opus audio with FFmpeg.
+   - Partial transcriptions appear immediately in light gray (the 'aperçu')
-   - **PyAudioWPatch**: Server receives raw PCM audio directly.
+   - Finalized text appears in normal color
-   - Audio is streamed into Whisper for transcription.
-3. **Real-time Output**:
-   - Partial transcriptions appear immediately in light gray (the 'aperçu').
-   - Finalized text appears in normal color.
    - (When enabled) Different speakers are identified and highlighted
 
 ## 🚀 Deployment Guide

setup.py

@@ -1,7 +1,7 @@
 from setuptools import setup, find_packages
 setup(
     name="whisperlivekit",
-    version="0.1.5",
+    version="0.1.9",
     description="Real-time, Fully Local Whisper's Speech-to-Text and Speaker Diarization",
     long_description=open("README.md", "r", encoding="utf-8").read(),
     long_description_content_type="text/markdown",
@@ -25,7 +25,6 @@ setup(
         "whisper-timestamped": ["whisper-timestamped"],
         "mlx-whisper": ["mlx-whisper"],
         "openai": ["openai"],
-        "pyaudiowpatch": ["PyAudioWPatch"],
     },
     package_data={
         'whisperlivekit': ['web/*.html'],

whisperlivekit/__init__.py

@@ -1,4 +1,5 @@
-from .core import WhisperLiveKit, _parse_args_internal, get_parsed_args
+from .core import TranscriptionEngine
 from .audio_processor import AudioProcessor
-
+from .web.web_interface import get_web_interface_html
-__all__ = ['WhisperLiveKit', 'AudioProcessor', '_parse_args_internal', 'get_parsed_args']
+from .parse_args import parse_args
+__all__ = ['TranscriptionEngine', 'AudioProcessor', 'get_web_interface_html', 'parse_args']

whisperlivekit/audio_processor.py

@@ -2,25 +2,20 @@ import asyncio
 import numpy as np
 import ffmpeg
 from time import time, sleep
-import platform  # To check OS
-
-try:
-    import pyaudiowpatch as pyaudio
-    PYAUDIOWPATCH_AVAILABLE = True
-except ImportError:
-    pyaudio = None
-    PYAUDIOWPATCH_AVAILABLE = False
 import math
 import logging
 import traceback
 from datetime import timedelta
 from whisperlivekit.timed_objects import ASRToken
 from whisperlivekit.whisper_streaming_custom.whisper_online import online_factory
-from whisperlivekit.core import WhisperLiveKit
+from whisperlivekit.core import TranscriptionEngine
 
 # Set up logging once
 logging.basicConfig(level=logging.INFO, format="%(asctime)s - %(levelname)s - %(message)s")
 logger = logging.getLogger(__name__)
+logger.setLevel(logging.DEBUG)
+
+SENTINEL = object() # unique sentinel object for end of stream marker
 
 def format_time(seconds: float) -> str:
     """Format seconds as HH:MM:SS."""
@@ -32,10 +27,13 @@ class AudioProcessor:
     Handles audio processing, state management, and result formatting.
     """
     
-    def __init__(self):
+    def __init__(self, **kwargs):
         """Initialize the audio processor with configuration, models, and state."""
         
-        models = WhisperLiveKit()
+        if 'transcription_engine' in kwargs and isinstance(kwargs['transcription_engine'], TranscriptionEngine):
+            models = kwargs['transcription_engine']
+        else:
+            models = TranscriptionEngine(**kwargs)
         
         # Audio processing settings
         self.args = models.args
@@ -48,8 +46,9 @@ class AudioProcessor:
         self.last_ffmpeg_activity = time()
         self.ffmpeg_health_check_interval = 5
         self.ffmpeg_max_idle_time = 10
- 
+
         # State management
+        self.is_stopping = False
         self.tokens = []
         self.buffer_transcription = ""
         self.buffer_diarization = ""
@@ -65,87 +64,55 @@ class AudioProcessor:
         self.asr = models.asr
         self.tokenizer = models.tokenizer
         self.diarization = models.diarization
+        self.ffmpeg_process = self.start_ffmpeg_decoder()
         self.transcription_queue = asyncio.Queue() if self.args.transcription else None
         self.diarization_queue = asyncio.Queue() if self.args.diarization else None
         self.pcm_buffer = bytearray()
-        self.ffmpeg_process = None
-        self.pyaudio_instance = None
-        self.pyaudio_stream = None
 
-        # Initialize audio input based on args
+        # Task references
-        if self.args.audio_input == "websocket":
+        self.transcription_task = None
-            self.ffmpeg_process = self.start_ffmpeg_decoder()
+        self.diarization_task = None
-        elif self.args.audio_input == "pyaudiowpatch":
+        self.ffmpeg_reader_task = None
-            if not PYAUDIOWPATCH_AVAILABLE:
+        self.watchdog_task = None
-                logger.error("PyAudioWPatch selected but not installed. Please install it: pip install whisperlivekit[pyaudiowpatch]")
+        self.all_tasks_for_cleanup = []
-                raise ImportError("PyAudioWPatch not found.")
-            if platform.system() != "Windows":
-                 logger.error("PyAudioWPatch is only supported on Windows.")
-                 raise OSError("PyAudioWPatch requires Windows.")
-            self.initialize_pyaudiowpatch()
-        else:
-            raise ValueError(f"Unsupported audio input type: {self.args.audio_input}")
         
         # Initialize transcription engine if enabled
         if self.args.transcription:
             self.online = online_factory(self.args, models.asr, models.tokenizer)
 
-    def initialize_pyaudiowpatch(self):
-        """Initialize PyAudioWPatch for audio input."""
-        logger.info("Initializing PyAudioWPatch...")
-        try:
-            self.pyaudio_instance = pyaudio.PyAudio()
-            # Find the default WASAPI loopback device
-            wasapi_info = self.pyaudio_instance.get_host_api_info_by_type(pyaudio.paWASAPI)
-            default_speakers = self.pyaudio_instance.get_device_info_by_index(wasapi_info["defaultOutputDevice"])
-
-            if not default_speakers["isLoopbackDevice"]:
-                for loopback in self.pyaudio_instance.get_loopback_device_info_generator():
-                    if default_speakers["name"] in loopback["name"]:
-                        default_speakers = loopback
-                        break
-                else:
-                    logger.error("Default loopback output device not found.")
-                    raise OSError("Default loopback output device not found.")
-
-            logger.info(f"Using loopback device: {default_speakers['name']}")
-            self.pyaudio_stream = self.pyaudio_instance.open(
-                format=pyaudio.paInt16,
-                channels=default_speakers["maxInputChannels"],
-                rate=int(default_speakers["defaultSampleRate"]),
-                input=True,
-                input_device_index=default_speakers["index"],
-                frames_per_buffer=int(self.sample_rate * self.args.min_chunk_size)
-            )
-            self.sample_rate = int(default_speakers["defaultSampleRate"])
-            self.channels = default_speakers["maxInputChannels"]
-            self.samples_per_sec = int(self.sample_rate * self.args.min_chunk_size)
-            self.bytes_per_sample = 2
-            self.bytes_per_sec = self.samples_per_sec * self.bytes_per_sample
-            logger.info(f"PyAudioWPatch initialized with {self.channels} channels and {self.sample_rate} Hz sample rate.")
-
-        except Exception as e:
-            logger.error(f"Failed to initialize PyAudioWPatch: {e}")
-            logger.error(traceback.format_exc())
-            if self.pyaudio_instance:
-                self.pyaudio_instance.terminate()
-            raise
-
     def convert_pcm_to_float(self, pcm_buffer):
         """Convert PCM buffer in s16le format to normalized NumPy array."""
-        if isinstance(pcm_buffer, (bytes, bytearray)):
+        return np.frombuffer(pcm_buffer, dtype=np.int16).astype(np.float32) / 32768.0
-            return np.frombuffer(pcm_buffer, dtype=np.int16).astype(np.float32) / 32768.0
-        else:
-            logger.error(f"Invalid buffer type for PCM conversion: {type(pcm_buffer)}")
-            return np.array([], dtype=np.float32)
-
 
     def start_ffmpeg_decoder(self):
         """Start FFmpeg process for WebM to PCM conversion."""
-        return (ffmpeg.input("pipe:0", format="webm")
+        try:
-                .output("pipe:1", format="s16le", acodec="pcm_s16le", 
+            return (ffmpeg.input("pipe:0", format="webm")
-                        ac=self.channels, ar=str(self.sample_rate))
+                    .output("pipe:1", format="s16le", acodec="pcm_s16le", 
-                .run_async(pipe_stdin=True, pipe_stdout=True, pipe_stderr=True))
+                            ac=self.channels, ar=str(self.sample_rate))
+                    .run_async(pipe_stdin=True, pipe_stdout=True, pipe_stderr=True))
+        except FileNotFoundError:
+            error = """
+            FFmpeg is not installed or not found in your system's PATH.
+            Please install FFmpeg to enable audio processing.
+
+            Installation instructions:
+
+            # Ubuntu/Debian:
+            sudo apt update && sudo apt install ffmpeg
+
+            # macOS (using Homebrew):
+            brew install ffmpeg
+
+            # Windows:
+            # 1. Download the latest static build from https://ffmpeg.org/download.html
+            # 2. Extract the archive (e.g., to C:\\FFmpeg).
+            # 3. Add the 'bin' directory (e.g., C:\\FFmpeg\\bin) to your system's PATH environment variable.
+
+            After installation, please restart the application.
+            """
+            logger.error(error)
+            raise FileNotFoundError(error)
 
     async def restart_ffmpeg(self):
         """Restart the FFmpeg process after failure."""
@@ -194,45 +161,6 @@ class AudioProcessor:
                 logger.critical(f"Failed to restart FFmpeg process on second attempt: {e2}")
                 logger.critical(traceback.format_exc())
 
-    async def pyaudiowpatch_reader(self):
-        """Read audio data from PyAudioWPatch stream and process it."""
-        logger.info("Starting PyAudioWPatch reader task.")
-        loop = asyncio.get_event_loop()
-
-        while True:
-            try:
-                chunk = await loop.run_in_executor(
-                    None,
-                    self.pyaudio_stream.read,
-                    int(self.sample_rate * self.args.min_chunk_size),
-                    False
-                )
-
-                if not chunk:
-                    logger.info("PyAudioWPatch stream closed or read empty chunk.")
-                    await asyncio.sleep(0.1)
-                    continue
-
-                pcm_array = self.convert_pcm_to_float(chunk)
-
-                if self.args.diarization and self.diarization_queue:
-                    await self.diarization_queue.put(pcm_array.copy())
-
-                if self.args.transcription and self.transcription_queue:
-                    await self.transcription_queue.put(pcm_array.copy())
-
-            except OSError as e:
-                 logger.error(f"PyAudioWPatch stream error: {e}")
-                 logger.error(traceback.format_exc())
-                 break
-            except Exception as e:
-                logger.error(f"Exception in pyaudiowpatch_reader: {e}")
-                logger.error(traceback.format_exc())
-                await asyncio.sleep(1) # Wait before retrying or breaking
-                break
-        logger.info("PyAudioWPatch reader task finished.")
-
-
     async def update_transcription(self, new_tokens, buffer, end_buffer, full_transcription, sep):
         """Thread-safe update of transcription with new data."""
         async with self.lock:
@@ -318,7 +246,7 @@ class AudioProcessor:
                     self.last_ffmpeg_activity = time()
                         
                 if not chunk:
-                    logger.info("FFmpeg stdout closed.")
+                    logger.info("FFmpeg stdout closed, no more data to read.")
                     break
                     
                 self.pcm_buffer.extend(chunk)
@@ -353,45 +281,86 @@ class AudioProcessor:
                 logger.warning(f"Exception in ffmpeg_stdout_reader: {e}")
                 logger.warning(f"Traceback: {traceback.format_exc()}")
                 break
+        
+        logger.info("FFmpeg stdout processing finished. Signaling downstream processors.")
+        if self.args.transcription and self.transcription_queue:
+            await self.transcription_queue.put(SENTINEL)
+            logger.debug("Sentinel put into transcription_queue.")
+        if self.args.diarization and self.diarization_queue:
+            await self.diarization_queue.put(SENTINEL)
+            logger.debug("Sentinel put into diarization_queue.")
+
 
     async def transcription_processor(self):
         """Process audio chunks for transcription."""
         self.full_transcription = ""
         self.sep = self.online.asr.sep
+        cumulative_pcm_duration_stream_time = 0.0
         
         while True:
             try:
                 pcm_array = await self.transcription_queue.get()
+                if pcm_array is SENTINEL:
+                    logger.debug("Transcription processor received sentinel. Finishing.")
+                    self.transcription_queue.task_done()
+                    break
                 
-                logger.info(f"{len(self.online.audio_buffer) / self.online.SAMPLING_RATE} seconds of audio to process.")
+                if not self.online: # Should not happen if queue is used
+                    logger.warning("Transcription processor: self.online not initialized.")
+                    self.transcription_queue.task_done()
+                    continue
+
+                asr_internal_buffer_duration_s = len(self.online.audio_buffer) / self.online.SAMPLING_RATE
+                transcription_lag_s = max(0.0, time() - self.beg_loop - self.end_buffer)
+
+                logger.info(
+                    f"ASR processing: internal_buffer={asr_internal_buffer_duration_s:.2f}s, "
+                    f"lag={transcription_lag_s:.2f}s."
+                )
                 
                 # Process transcription
-                self.online.insert_audio_chunk(pcm_array)
+                duration_this_chunk = len(pcm_array) / self.sample_rate if isinstance(pcm_array, np.ndarray) else 0
-                new_tokens = self.online.process_iter()
+                cumulative_pcm_duration_stream_time += duration_this_chunk
+                stream_time_end_of_current_pcm = cumulative_pcm_duration_stream_time
+
+                self.online.insert_audio_chunk(pcm_array, stream_time_end_of_current_pcm)
+                new_tokens, current_audio_processed_upto = self.online.process_iter()
                 
                 if new_tokens:
                     self.full_transcription += self.sep.join([t.text for t in new_tokens])
                     
                 # Get buffer information
-                _buffer = self.online.get_buffer()
+                _buffer_transcript_obj = self.online.get_buffer()
-                buffer = _buffer.text
+                buffer_text = _buffer_transcript_obj.text
-                end_buffer = _buffer.end if _buffer.end else (
+                
-                    new_tokens[-1].end if new_tokens else 0
+                candidate_end_times = [self.end_buffer]
-                )
+
+                if new_tokens:
+                    candidate_end_times.append(new_tokens[-1].end)
+                
+                if _buffer_transcript_obj.end is not None:
+                    candidate_end_times.append(_buffer_transcript_obj.end)
+                
+                candidate_end_times.append(current_audio_processed_upto)
+                
+                new_end_buffer = max(candidate_end_times)
                 
                 # Avoid duplicating content
-                if buffer in self.full_transcription:
+                if buffer_text in self.full_transcription:
-                    buffer = ""
+                    buffer_text = ""
                     
                 await self.update_transcription(
-                    new_tokens, buffer, end_buffer, self.full_transcription, self.sep
+                    new_tokens, buffer_text, new_end_buffer, self.full_transcription, self.sep
                 )
+                self.transcription_queue.task_done()
                 
             except Exception as e:
                 logger.warning(f"Exception in transcription_processor: {e}")
                 logger.warning(f"Traceback: {traceback.format_exc()}")
-            finally:
+                if 'pcm_array' in locals() and pcm_array is not SENTINEL : # Check if pcm_array was assigned from queue
-                self.transcription_queue.task_done()
+                    self.transcription_queue.task_done()
+        logger.info("Transcription processor task finished.")
+
 
     async def diarization_processor(self, diarization_obj):
         """Process audio chunks for speaker diarization."""
@@ -400,23 +369,33 @@ class AudioProcessor:
         while True:
             try:
                 pcm_array = await self.diarization_queue.get()
+                if pcm_array is SENTINEL:
+                    logger.debug("Diarization processor received sentinel. Finishing.")
+                    self.diarization_queue.task_done()
+                    break
                 
                 # Process diarization
                 await diarization_obj.diarize(pcm_array)
                 
-                # Get current state and update speakers
+                async with self.lock:
-                state = await self.get_current_state()
+                    new_end = diarization_obj.assign_speakers_to_tokens(
-                new_end = diarization_obj.assign_speakers_to_tokens(
+                        self.end_attributed_speaker, 
-                    state["end_attributed_speaker"], state["tokens"]
+                        self.tokens,
-                )
+                        use_punctuation_split=self.args.punctuation_split
+                    )
+                    self.end_attributed_speaker = new_end
+                    if buffer_diarization:
+                        self.buffer_diarization = buffer_diarization
                 
-                await self.update_diarization(new_end, buffer_diarization)
+                self.diarization_queue.task_done()
                 
             except Exception as e:
                 logger.warning(f"Exception in diarization_processor: {e}")
                 logger.warning(f"Traceback: {traceback.format_exc()}")
-            finally:
+                if 'pcm_array' in locals() and pcm_array is not SENTINEL:
-                self.diarization_queue.task_done()
+                    self.diarization_queue.task_done()
+        logger.info("Diarization processor task finished.")
+
 
     async def results_formatter(self):
         """Format processing results for output."""
@@ -480,31 +459,51 @@ class AudioProcessor:
                     await self.update_diarization(end_attributed_speaker, combined)
                     buffer_diarization = combined
                 
-                # Create response object
+                response_status = "active_transcription"
-                if not lines:
+                final_lines_for_response = lines.copy()
-                    lines = [{
+
+                if not tokens and not buffer_transcription and not buffer_diarization:
+                    response_status = "no_audio_detected"
+                    final_lines_for_response = []
+                elif response_status == "active_transcription" and not final_lines_for_response:
+                    final_lines_for_response = [{
                         "speaker": 1,
                         "text": "",
-                        "beg": format_time(0),
+                        "beg": format_time(state.get("end_buffer", 0)),
-                        "end": format_time(tokens[-1].end if tokens else 0),
+                        "end": format_time(state.get("end_buffer", 0)),
                         "diff": 0
                     }]
                 
                 response = {
-                    "lines": lines, 
+                    "status": response_status,
+                    "lines": final_lines_for_response,
                     "buffer_transcription": buffer_transcription,
                     "buffer_diarization": buffer_diarization,
                     "remaining_time_transcription": state["remaining_time_transcription"],
                     "remaining_time_diarization": state["remaining_time_diarization"]
                 }
                 
-                # Only yield if content has changed
+                current_response_signature = f"{response_status} | " + \
-                response_content = ' '.join([f"{line['speaker']} {line['text']}" for line in lines]) + \
+                                           ' '.join([f"{line['speaker']} {line['text']}" for line in final_lines_for_response]) + \
-                                  f" | {buffer_transcription} | {buffer_diarization}"
+                                           f" | {buffer_transcription} | {buffer_diarization}"
                 
-                if response_content != self.last_response_content and (lines or buffer_transcription or buffer_diarization):
+                if current_response_signature != self.last_response_content and \
+                   (final_lines_for_response or buffer_transcription or buffer_diarization or response_status == "no_audio_detected"):
                     yield response
-                    self.last_response_content = response_content
+                    self.last_response_content = current_response_signature
+                
+                # Check for termination condition
+                if self.is_stopping:
+                    all_processors_done = True
+                    if self.args.transcription and self.transcription_task and not self.transcription_task.done():
+                        all_processors_done = False
+                    if self.args.diarization and self.diarization_task and not self.diarization_task.done():
+                        all_processors_done = False
+                    
+                    if all_processors_done:
+                        logger.info("Results formatter: All upstream processors are done and in stopping state. Terminating.")
+                        final_state = await self.get_current_state()
+                        return
                 
                 await asyncio.sleep(0.1)  # Avoid overwhelming the client
                 
@@ -515,85 +514,117 @@ class AudioProcessor:
         
     async def create_tasks(self):
         """Create and start processing tasks."""
-            
+        self.all_tasks_for_cleanup = []
-        tasks = []    
+        processing_tasks_for_watchdog = []
+
         if self.args.transcription and self.online:
-            tasks.append(asyncio.create_task(self.transcription_processor()))
+            self.transcription_task = asyncio.create_task(self.transcription_processor())
-
+            self.all_tasks_for_cleanup.append(self.transcription_task)
+            processing_tasks_for_watchdog.append(self.transcription_task)
+            
         if self.args.diarization and self.diarization:
-            tasks.append(asyncio.create_task(self.diarization_processor(self.diarization))) # Corrected indentation
+            self.diarization_task = asyncio.create_task(self.diarization_processor(self.diarization))
-
+            self.all_tasks_for_cleanup.append(self.diarization_task)
-        if self.args.audio_input == "websocket":
+            processing_tasks_for_watchdog.append(self.diarization_task)
-            tasks.append(asyncio.create_task(self.ffmpeg_stdout_reader()))
+        
-        elif self.args.audio_input == "pyaudiowpatch":
+        self.ffmpeg_reader_task = asyncio.create_task(self.ffmpeg_stdout_reader())
-            tasks.append(asyncio.create_task(self.pyaudiowpatch_reader()))
+        self.all_tasks_for_cleanup.append(self.ffmpeg_reader_task)
+        processing_tasks_for_watchdog.append(self.ffmpeg_reader_task)
 
         # Monitor overall system health
-        async def watchdog():
+        self.watchdog_task = asyncio.create_task(self.watchdog(processing_tasks_for_watchdog))
-            while True:
+        self.all_tasks_for_cleanup.append(self.watchdog_task)
-                try:
-                    await asyncio.sleep(10)  # Check every 10 seconds instead of 60
-                    
-                    current_time = time()
-                    # Check for stalled tasks
-                    for i, task in enumerate(tasks):
-                        if task.done():
-                            exc = task.exception() if task.done() else None
-                            task_name = task.get_name() if hasattr(task, 'get_name') else f"Task {i}"
-                            logger.error(f"{task_name} unexpectedly completed with exception: {exc}")
-                    
-                    if self.args.audio_input == "websocket":
-                        ffmpeg_idle_time = current_time - self.last_ffmpeg_activity
-                        if ffmpeg_idle_time > 15:  # 15 seconds instead of 180
-                            logger.warning(f"FFmpeg idle for {ffmpeg_idle_time:.2f}s - may need attention")
-
-                            # Force restart after 30 seconds of inactivity (instead of 600)
-                            if ffmpeg_idle_time > 30:
-                                logger.error("FFmpeg idle for too long, forcing restart")
-                                await self.restart_ffmpeg()
-
-                    elif self.args.audio_input == "pyaudiowpatch":
-                         if self.pyaudio_stream and not self.pyaudio_stream.is_active():
-                              logger.warning("PyAudioWPatch stream is not active. Attempting to restart or handle.")
-
-                except Exception as e:
-                    logger.error(f"Error in watchdog task: {e}")
-                    logger.error(traceback.format_exc())
-
-        tasks.append(asyncio.create_task(watchdog()))
-        self.tasks = tasks
         
         return self.results_formatter()
+
+    async def watchdog(self, tasks_to_monitor):
+        """Monitors the health of critical processing tasks."""
+        while True:
+            try:
+                await asyncio.sleep(10)
+                current_time = time()
+
+                for i, task in enumerate(tasks_to_monitor):
+                    if task.done():
+                        exc = task.exception()
+                        task_name = task.get_name() if hasattr(task, 'get_name') else f"Monitored Task {i}"
+                        if exc:
+                            logger.error(f"{task_name} unexpectedly completed with exception: {exc}")
+                        else:
+                            logger.info(f"{task_name} completed normally.")
+                
+                ffmpeg_idle_time = current_time - self.last_ffmpeg_activity
+                if ffmpeg_idle_time > 15:
+                    logger.warning(f"FFmpeg idle for {ffmpeg_idle_time:.2f}s - may need attention.")
+                    if ffmpeg_idle_time > 30 and not self.is_stopping:
+                        logger.error("FFmpeg idle for too long and not in stopping phase, forcing restart.")
+                        await self.restart_ffmpeg()
+            except asyncio.CancelledError:
+                logger.info("Watchdog task cancelled.")
+                break
+            except Exception as e:
+                logger.error(f"Error in watchdog task: {e}", exc_info=True)
         
     async def cleanup(self):
         """Clean up resources when processing is complete."""
-        for task in self.tasks:
+        logger.info("Starting cleanup of AudioProcessor resources.")
-            task.cancel()
+        for task in self.all_tasks_for_cleanup:
-            
+            if task and not task.done():
-        try:
+                task.cancel()
-            await asyncio.gather(*self.tasks, return_exceptions=True)
+        
-            if self.args.audio_input == "websocket" and self.ffmpeg_process:
+        created_tasks = [t for t in self.all_tasks_for_cleanup if t]
-                if self.ffmpeg_process.stdin:
+        if created_tasks:
+            await asyncio.gather(*created_tasks, return_exceptions=True)
+        logger.info("All processing tasks cancelled or finished.")
+
+        if self.ffmpeg_process:
+            if self.ffmpeg_process.stdin and not self.ffmpeg_process.stdin.closed:
+                try:
                     self.ffmpeg_process.stdin.close()
-                if self.ffmpeg_process.poll() is None:
+                except Exception as e:
-                     self.ffmpeg_process.wait()
+                    logger.warning(f"Error closing ffmpeg stdin during cleanup: {e}")
-            elif self.args.audio_input == "pyaudiowpatch":
-                if self.pyaudio_stream:
-                    self.pyaudio_stream.stop_stream()
-                    self.pyaudio_stream.close()
-                    logger.info("PyAudioWPatch stream closed.")
-                if self.pyaudio_instance:
-                    self.pyaudio_instance.terminate()
-                    logger.info("PyAudioWPatch instance terminated.")
-        except Exception as e:
-            logger.warning(f"Error during cleanup: {e}")
-            logger.warning(traceback.format_exc())
             
-        if self.args.diarization and hasattr(self, 'diarization'):
+            # Wait for ffmpeg process to terminate
+            if self.ffmpeg_process.poll() is None: # Check if process is still running
+                logger.info("Waiting for FFmpeg process to terminate...")
+                try:
+                    # Run wait in executor to avoid blocking async loop
+                    await asyncio.get_event_loop().run_in_executor(None, self.ffmpeg_process.wait, 5.0) # 5s timeout
+                except Exception as e: # subprocess.TimeoutExpired is not directly caught by asyncio.wait_for with run_in_executor
+                    logger.warning(f"FFmpeg did not terminate gracefully, killing. Error: {e}")
+                    self.ffmpeg_process.kill()
+                    await asyncio.get_event_loop().run_in_executor(None, self.ffmpeg_process.wait) # Wait for kill
+            logger.info("FFmpeg process terminated.")
+
+        if self.args.diarization and hasattr(self, 'diarization') and hasattr(self.diarization, 'close'):
             self.diarization.close()
+        logger.info("AudioProcessor cleanup complete.")
+
 
     async def process_audio(self, message):
         """Process incoming audio data."""
+        # If already stopping or stdin is closed, ignore further audio, especially residual chunks.
+        if self.is_stopping or (self.ffmpeg_process and self.ffmpeg_process.stdin and self.ffmpeg_process.stdin.closed):
+            logger.warning(f"AudioProcessor is stopping or stdin is closed. Ignoring incoming audio message (length: {len(message)}).")
+            if not message and self.ffmpeg_process and self.ffmpeg_process.stdin and not self.ffmpeg_process.stdin.closed:
+                 logger.info("Received empty message while already in stopping state; ensuring stdin is closed.")
+                 try:
+                    self.ffmpeg_process.stdin.close()
+                 except Exception as e:
+                    logger.warning(f"Error closing ffmpeg stdin on redundant stop signal during stopping state: {e}")
+            return
+
+        if not message: # primary signal to start stopping
+            logger.info("Empty audio message received, initiating stop sequence.")
+            self.is_stopping = True
+            if self.ffmpeg_process and self.ffmpeg_process.stdin and not self.ffmpeg_process.stdin.closed:
+                try:
+                    self.ffmpeg_process.stdin.close()
+                    logger.info("FFmpeg stdin closed due to primary stop signal.")
+                except Exception as e:
+                    logger.warning(f"Error closing ffmpeg stdin on stop: {e}")
+            return
+
         retry_count = 0
         max_retries = 3
         
@@ -602,37 +633,14 @@ class AudioProcessor:
         if not hasattr(self, '_last_heartbeat') or current_time - self._last_heartbeat >= 10:
             logger.debug(f"Processing audio chunk, last FFmpeg activity: {current_time - self.last_ffmpeg_activity:.2f}s ago")
             self._last_heartbeat = current_time
-
+        
-        if self.args.audio_input != "websocket":
-            # logger.debug("Audio input is not WebSocket, skipping process_audio.")
-            return # Do nothing if input is not WebSocket
-
         while retry_count < max_retries:
             try:
-
+                if not self.ffmpeg_process or not hasattr(self.ffmpeg_process, 'stdin') or self.ffmpeg_process.poll() is not None:
-                if not self.ffmpeg_process or self.ffmpeg_process.poll() is not None:
+                    logger.warning("FFmpeg process not available, restarting...")
-                    logger.warning("FFmpeg process not running or unavailable, attempting restart...")
                     await self.restart_ffmpeg()
-
+                
-                    if not self.ffmpeg_process or self.ffmpeg_process.poll() is not None:
+                loop = asyncio.get_running_loop()                
-                         logger.error("FFmpeg restart failed or process terminated immediately.")
-                         # maybe raise an error or break after retries
-                         await asyncio.sleep(1)
-                         retry_count += 1
-                         continue
-
-                # Ensure stdin is available
-                if not hasattr(self.ffmpeg_process, 'stdin') or self.ffmpeg_process.stdin.closed:
-                     logger.warning("FFmpeg stdin is not available or closed. Restarting...")
-                     await self.restart_ffmpeg()
-                     if not hasattr(self.ffmpeg_process, 'stdin') or self.ffmpeg_process.stdin.closed:
-                          logger.error("FFmpeg stdin still unavailable after restart.")
-                          await asyncio.sleep(1)
-                          retry_count += 1
-                          continue
-
-
-                loop = asyncio.get_running_loop()
                 try:
                     await asyncio.wait_for(
                         loop.run_in_executor(None, lambda: self.ffmpeg_process.stdin.write(message)),
@@ -668,4 +676,4 @@ class AudioProcessor:
                 else:
                     logger.error("Maximum retries reached for FFmpeg process")
                     await self.restart_ffmpeg()
-                    return
+                    return

whisperlivekit/basic_server.py

@@ -2,48 +2,26 @@ from contextlib import asynccontextmanager
 from fastapi import FastAPI, WebSocket, WebSocketDisconnect
 from fastapi.responses import HTMLResponse
 from fastapi.middleware.cors import CORSMiddleware
-
+from whisperlivekit import TranscriptionEngine, AudioProcessor, get_web_interface_html, parse_args
-from whisperlivekit import WhisperLiveKit, get_parsed_args
-from whisperlivekit.audio_processor import AudioProcessor
-
 import asyncio
 import logging
-import os, sys
 
 logging.basicConfig(level=logging.INFO, format="%(asctime)s - %(levelname)s - %(message)s")
 logging.getLogger().setLevel(logging.WARNING)
 logger = logging.getLogger(__name__)
 logger.setLevel(logging.DEBUG)
 
+args = parse_args()
+transcription_engine = None
+
 @asynccontextmanager
 async def lifespan(app: FastAPI):
-    logger.info("Starting up...")
+    global transcription_engine
-    kit = WhisperLiveKit()
+    transcription_engine = TranscriptionEngine(
-    app.state.kit = kit
+        **vars(args),
-    logger.info(f"Audio Input mode: {kit.args.audio_input}")
+    )
-
-    audio_processor = AudioProcessor()
-    app.state.audio_processor = audio_processor
-    app.state.results_generator = None # Initialize
-
-    if kit.args.audio_input == "pyaudiowpatch":
-        logger.info("Starting PyAudioWPatch processing tasks...")
-        try:
-            app.state.results_generator = await audio_processor.create_tasks()
-        except Exception as e:
-             logger.critical(f"Failed to start PyAudioWPatch processing: {e}", exc_info=True)
-    else:
-        logger.info("WebSocket input mode selected. Processing will start on client connection.")
-
     yield
 
-    logger.info("Shutting down...")
-    if hasattr(app.state, 'audio_processor') and app.state.audio_processor:
-        logger.info("Cleaning up AudioProcessor...")
-        await app.state.audio_processor.cleanup()
-    logger.info("Shutdown complete.")
-
-
 app = FastAPI(lifespan=lifespan)
 app.add_middleware(
     CORSMiddleware,
@@ -53,126 +31,74 @@ app.add_middleware(
     allow_headers=["*"],
 )
 
-
 @app.get("/")
 async def get():
-    return HTMLResponse(app.state.kit.web_interface())
+    return HTMLResponse(get_web_interface_html())
 
 
-async def handle_websocket_results(websocket: WebSocket, results_generator):
+async def handle_websocket_results(websocket, results_generator):
     """Consumes results from the audio processor and sends them via WebSocket."""
     try:
         async for response in results_generator:
             await websocket.send_json(response)
+        # when the results_generator finishes it means all audio has been processed
+        logger.info("Results generator finished. Sending 'ready_to_stop' to client.")
+        await websocket.send_json({"type": "ready_to_stop"})
+    except WebSocketDisconnect:
+        logger.info("WebSocket disconnected while handling results (client likely closed connection).")
     except Exception as e:
         logger.warning(f"Error in WebSocket results handler: {e}")
 
 
 @app.websocket("/asr")
 async def websocket_endpoint(websocket: WebSocket):
+    global transcription_engine
+    audio_processor = AudioProcessor(
+        transcription_engine=transcription_engine,
+    )
     await websocket.accept()
-    logger.info("WebSocket connection accepted.")
+    logger.info("WebSocket connection opened.")
-
+            
-    audio_processor = app.state.audio_processor
+    results_generator = await audio_processor.create_tasks()
-    kit_args = app.state.kit.args
+    websocket_task = asyncio.create_task(handle_websocket_results(websocket, results_generator))
-    results_generator = None
-    websocket_task = None
-    receive_task = None
 
     try:
-        if kit_args.audio_input == "websocket":
+        while True:
-            logger.info("WebSocket mode: Starting processing tasks for this connection.")
+            message = await websocket.receive_bytes()
-            results_generator = await audio_processor.create_tasks()
+            await audio_processor.process_audio(message)
-            websocket_task = asyncio.create_task(handle_websocket_results(websocket, results_generator))
+    except KeyError as e:
-
+        if 'bytes' in str(e):
-            async def receive_audio():
+            logger.warning(f"Client has closed the connection.")
-                try:
-                    while True:
-                        message = await websocket.receive_bytes()
-                        await audio_processor.process_audio(message)
-                except WebSocketDisconnect:
-                    logger.info("WebSocket disconnected by client (receive_audio).")
-                except Exception as e:
-                    logger.error(f"Error receiving audio: {e}", exc_info=True)
-                finally:
-                     logger.debug("Receive audio task finished.")
-
-
-            receive_task = asyncio.create_task(receive_audio())
-            done, pending = await asyncio.wait(
-                {websocket_task, receive_task},
-                return_when=asyncio.FIRST_COMPLETED,
-            )
-            for task in pending:
-                task.cancel() # Cancel the other task
-
-        elif kit_args.audio_input == "pyaudiowpatch":
-            logger.info("PyAudioWPatch mode: Streaming existing results.")
-            results_generator = app.state.results_generator
-            if results_generator is None:
-                 logger.error("PyAudioWPatch results generator not available. Was startup successful?")
-                 await websocket.close(code=1011, reason="Server error: Audio processing not started.")
-                 return
-
-            websocket_task = asyncio.create_task(handle_websocket_results(websocket, results_generator))
-            await websocket_task
-
         else:
-             logger.error(f"Unsupported audio input mode configured: {kit_args.audio_input}")
+            logger.error(f"Unexpected KeyError in websocket_endpoint: {e}", exc_info=True)
-             await websocket.close(code=1011, reason="Server configuration error.")
-
     except WebSocketDisconnect:
-        logger.info("WebSocket disconnected by client.")
+        logger.info("WebSocket disconnected by client during message receiving loop.")
     except Exception as e:
-        logger.error(f"Error in WebSocket endpoint: {e}", exc_info=True)
+        logger.error(f"Unexpected error in websocket_endpoint main loop: {e}", exc_info=True)
-        # Attempt to close gracefully
-        try:
-            await websocket.close(code=1011, reason=f"Server error: {e}")
-        except Exception:
-            pass # Ignore errors during close after another error
     finally:
-        logger.info("Cleaning up WebSocket connection...")
+        logger.info("Cleaning up WebSocket endpoint...")
-        if websocket_task and not websocket_task.done():
+        if not websocket_task.done():
             websocket_task.cancel()
-        if receive_task and not receive_task.done():
+        try:
-            receive_task.cancel()
+            await websocket_task
-
+        except asyncio.CancelledError:
-        if kit_args.audio_input == "websocket":
+            logger.info("WebSocket results handler task was cancelled.")
-             pass
+        except Exception as e:
-
+            logger.warning(f"Exception while awaiting websocket_task completion: {e}")
-        logger.info("WebSocket connection closed.")
+            
+        await audio_processor.cleanup()
+        logger.info("WebSocket endpoint cleaned up successfully.")
 
 def main():
     """Entry point for the CLI command."""
     import uvicorn
-
+    
-    # Get the globally parsed arguments
-    args = get_parsed_args()
-
-    # Set logger level based on args
-    log_level_name = args.log_level.upper()
-    # Ensure the level name is valid for the logging module
-    numeric_level = getattr(logging, log_level_name, None)
-    if not isinstance(numeric_level, int):
-        logging.warning(f"Invalid log level: {args.log_level}. Defaulting to INFO.")
-        numeric_level = logging.INFO
-    logging.getLogger().setLevel(numeric_level) # Set root logger level
-    # Set our specific logger level too
-    logger.setLevel(numeric_level)
-    logger.info(f"Log level set to: {log_level_name}")
-
-    # Determine uvicorn log level (map CRITICAL to critical, etc.)
-    uvicorn_log_level = log_level_name.lower()
-    if uvicorn_log_level == "debug": # Uvicorn uses 'trace' for more verbose than debug
-         uvicorn_log_level = "trace"
-
-
     uvicorn_kwargs = {
         "app": "whisperlivekit.basic_server:app",
         "host":args.host, 
-        "port":args.port,
+        "port":args.port, 
         "reload": False,
-        "log_level": uvicorn_log_level,
+        "log_level": "info",
         "lifespan": "on",
     }
     
@@ -185,7 +111,6 @@ def main():
             "ssl_keyfile": args.ssl_keyfile
         }
 
-
     if ssl_kwargs:
         uvicorn_kwargs = {**uvicorn_kwargs, **ssl_kwargs}
 

whisperlivekit/core.py

@@ -1,187 +1,66 @@
-import sys
-from argparse import Namespace, ArgumentParser
 try:
     from whisperlivekit.whisper_streaming_custom.whisper_online import backend_factory, warmup_asr
 except ImportError:
-    if '.' not in sys.path:
+    from .whisper_streaming_custom.whisper_online import backend_factory, warmup_asr
-        sys.path.insert(0, '.')
+from argparse import Namespace
-    from whisperlivekit.whisper_streaming_custom.whisper_online import backend_factory, warmup_asr
-
-def _parse_args_internal():
-    parser = ArgumentParser(description="Whisper FastAPI Online Server")
-    parser.add_argument(
-        "--host",
-        type=str,
-        default="localhost",
-        help="The host address to bind the server to.",
-    )
-    parser.add_argument(
-        "--port", type=int, default=8000, help="The port number to bind the server to."
-    )
-    parser.add_argument(
-        "--warmup-file",
-        type=str,
-        default=None,
-        dest="warmup_file",
-        help="""
-        The path to a speech audio wav file to warm up Whisper so that the very first chunk processing is fast.
-        If not set, uses https://github.com/ggerganov/whisper.cpp/raw/master/samples/jfk.wav.
-        If False, no warmup is performed.
-        """,
-    )
-
-    parser.add_argument(
-        "--confidence-validation",
-        action="store_true",
-        help="Accelerates validation of tokens using confidence scores. Transcription will be faster but punctuation might be less accurate.",
-    )
-
-    parser.add_argument(
-        "--diarization",
-        action="store_true",
-        default=False,
-        help="Enable speaker diarization.",
-    )
-
-    parser.add_argument(
-        "--no-transcription",
-        action="store_true",
-        help="Disable transcription to only see live diarization results.",
-    )
-    
-    parser.add_argument(
-        "--min-chunk-size",
-        type=float,
-        default=0.5,
-        help="Minimum audio chunk size in seconds. It waits up to this time to do processing. If the processing takes shorter time, it waits, otherwise it processes the whole segment that was received by this time.",
-    )
-    
-    parser.add_argument(
-        "--model",
-        type=str,
-        default="tiny",
-        help="Name size of the Whisper model to use (default: tiny). Suggested values: tiny.en,tiny,base.en,base,small.en,small,medium.en,medium,large-v1,large-v2,large-v3,large,large-v3-turbo. The model is automatically downloaded from the model hub if not present in model cache dir.",
-    )
-    
-    parser.add_argument(
-        "--model_cache_dir",
-        type=str,
-        default=None,
-        help="Overriding the default model cache dir where models downloaded from the hub are saved",
-    )
-    parser.add_argument(
-        "--model_dir",
-        type=str,
-        default=None,
-        help="Dir where Whisper model.bin and other files are saved. This option overrides --model and --model_cache_dir parameter.",
-    )
-    parser.add_argument(
-        "--lan",
-        "--language",
-        type=str,
-        default="auto",
-        help="Source language code, e.g. en,de,cs, or 'auto' for language detection.",
-    )
-    parser.add_argument(
-        "--task",
-        type=str,
-        default="transcribe",
-        choices=["transcribe", "translate"],
-        help="Transcribe or translate.",
-    )
-    parser.add_argument(
-        "--backend",
-        type=str,
-        default="faster-whisper",
-        choices=["faster-whisper", "whisper_timestamped", "mlx-whisper", "openai-api"],
-        help="Load only this backend for Whisper processing.",
-    )
-    parser.add_argument(
-        "--vac",
-        action="store_true",
-        default=False,
-        help="Use VAC = voice activity controller. Recommended. Requires torch.",
-    )
-    parser.add_argument(
-        "--vac-chunk-size", type=float, default=0.04, help="VAC sample size in seconds."
-    )
-
-    parser.add_argument(
-        "--no-vad",
-        action="store_true",
-        help="Disable VAD (voice activity detection).",
-    )
-    
-    parser.add_argument(
-        "--buffer_trimming",
-        type=str,
-        default="segment",
-        choices=["sentence", "segment"],
-        help='Buffer trimming strategy -- trim completed sentences marked with punctuation mark and detected by sentence segmenter, or the completed segments returned by Whisper. Sentence segmenter must be installed for "sentence" option.',
-    )
-    parser.add_argument(
-        "--buffer_trimming_sec",
-        type=float,
-        default=15,
-        help="Buffer trimming length threshold in seconds. If buffer length is longer, trimming sentence/segment is triggered.",
-    )
-    parser.add_argument(
-        "-l",
-        "--log-level",
-        dest="log_level",
-        choices=["DEBUG", "INFO", "WARNING", "ERROR", "CRITICAL"],
-        help="Set the log level",
-        default="DEBUG",
-    )
-    parser.add_argument(
-        "--audio-input",
-        type=str,
-        default="websocket",
-        choices=["websocket", "pyaudiowpatch"],
-        help="Source of the audio input. 'websocket' expects audio via WebSocket (default). 'pyaudiowpatch' uses PyAudioWPatch to capture system audio output.",
-    )
-    parser.add_argument("--ssl-certfile", type=str, help="Path to the SSL certificate file.", default=None)
-    parser.add_argument("--ssl-keyfile", type=str, help="Path to the SSL private key file.", default=None)
 
 
-    args = parser.parse_args()
+class TranscriptionEngine:
-
-    args.transcription = not args.no_transcription
-    args.vad = not args.no_vad
-    delattr(args, 'no_transcription')
-    delattr(args, 'no_vad')
-
-    return args
-
-_cli_args = _parse_args_internal()
-
-def get_parsed_args() -> Namespace:
-    """Returns the globally parsed command-line arguments."""
-    return _cli_args
-
-# --- WhisperLiveKit Class ---
-class WhisperLiveKit:
     _instance = None
     _initialized = False
-
+    
-    def __new__(cls, args: Namespace = None, **kwargs):
+    def __new__(cls, *args, **kwargs):
         if cls._instance is None:
             cls._instance = super().__new__(cls)
         return cls._instance
-
+    
-    def __init__(self, args: Namespace = None, **kwargs):
+    def __init__(self, **kwargs):
-        """
+        if TranscriptionEngine._initialized:
-        Initializes WhisperLiveKit.
-
-        Args:
-            args (Namespace, optional): Pre-parsed arguments. If None, uses globally parsed args.
-                                        Defaults to None.
-            **kwargs: Additional keyword arguments (currently not used directly but captured).
-        """
-        if WhisperLiveKit._initialized:
             return
 
-        self.args = args if args is not None else get_parsed_args()
+        defaults = {
+            "host": "localhost",
+            "port": 8000,
+            "warmup_file": None,
+            "confidence_validation": False,
+            "diarization": False,
+            "punctuation_split": False,
+            "min_chunk_size": 0.5,
+            "model": "tiny",
+            "model_cache_dir": None,
+            "model_dir": None,
+            "lan": "auto",
+            "task": "transcribe",
+            "backend": "faster-whisper",
+            "vac": False,
+            "vac_chunk_size": 0.04,
+            "buffer_trimming": "segment",
+            "buffer_trimming_sec": 15,
+            "log_level": "DEBUG",
+            "ssl_certfile": None,
+            "ssl_keyfile": None,
+            "transcription": True,
+            "vad": True,
+            "segmentation_model": "pyannote/segmentation-3.0",
+            "embedding_model": "pyannote/embedding",
+        }
+
+        config_dict = {**defaults, **kwargs}
+
+        if 'no_transcription' in kwargs:
+            config_dict['transcription'] = not kwargs['no_transcription']
+        if 'no_vad' in kwargs:
+            config_dict['vad'] = not kwargs['no_vad']
+        
+        config_dict.pop('no_transcription', None)
+        config_dict.pop('no_vad', None)
+
+        if 'language' in kwargs:
+            config_dict['lan'] = kwargs['language']
+        config_dict.pop('language', None) 
+
+        self.args = Namespace(**config_dict)
+        
         self.asr = None
         self.tokenizer = None
         self.diarization = None
@@ -192,13 +71,10 @@ class WhisperLiveKit:
 
         if self.args.diarization:
             from whisperlivekit.diarization.diarization_online import DiartDiarization
-            self.diarization = DiartDiarization()
+            self.diarization = DiartDiarization(
+                block_duration=self.args.min_chunk_size,
+                segmentation_model_name=self.args.segmentation_model,
+                embedding_model_name=self.args.embedding_model
+            )
             
-        WhisperLiveKit._initialized = True
+        TranscriptionEngine._initialized = True
-
-    def web_interface(self):
-        import pkg_resources
-        html_path = pkg_resources.resource_filename('whisperlivekit', 'web/live_transcription.html')
-        with open(html_path, "r", encoding="utf-8") as f:
-            html = f.read()
-        return html

whisperlivekit/diarization/diarization_online.py

@@ -3,7 +3,8 @@ import re
 import threading
 import numpy as np
 import logging
-
+import time
+from queue import SimpleQueue, Empty
 
 from diart import SpeakerDiarization, SpeakerDiarizationConfig
 from diart.inference import StreamingInference
@@ -13,6 +14,7 @@ from diart.sources import MicrophoneAudioSource
 from rx.core import Observer
 from typing import Tuple, Any, List
 from pyannote.core import Annotation
+import diart.models as m
 
 logger = logging.getLogger(__name__)
 
@@ -78,40 +80,114 @@ class DiarizationObserver(Observer):
 
 class WebSocketAudioSource(AudioSource):
     """
-    Custom AudioSource that blocks in read() until close() is called.
+    Buffers incoming audio and releases it in fixed-size chunks at regular intervals.
-    Use push_audio() to inject PCM chunks.
     """
-    def __init__(self, uri: str = "websocket", sample_rate: int = 16000):
+    def __init__(self, uri: str = "websocket", sample_rate: int = 16000, block_duration: float = 0.5):
         super().__init__(uri, sample_rate)
+        self.block_duration = block_duration
+        self.block_size = int(np.rint(block_duration * sample_rate))
+        self._queue = SimpleQueue()
+        self._buffer = np.array([], dtype=np.float32)
+        self._buffer_lock = threading.Lock()
         self._closed = False
         self._close_event = threading.Event()
+        self._processing_thread = None
+        self._last_chunk_time = time.time()
 
     def read(self):
+        """Start processing buffered audio and emit fixed-size chunks."""
+        self._processing_thread = threading.Thread(target=self._process_chunks)
+        self._processing_thread.daemon = True
+        self._processing_thread.start()
+        
         self._close_event.wait()
+        if self._processing_thread:
+            self._processing_thread.join(timeout=2.0)
+
+    def _process_chunks(self):
+        """Process audio from queue and emit fixed-size chunks at regular intervals."""
+        while not self._closed:
+            try:
+                audio_chunk = self._queue.get(timeout=0.1)
+                
+                with self._buffer_lock:
+                    self._buffer = np.concatenate([self._buffer, audio_chunk])
+                    
+                    while len(self._buffer) >= self.block_size:
+                        chunk = self._buffer[:self.block_size]
+                        self._buffer = self._buffer[self.block_size:]
+                        
+                        current_time = time.time()
+                        time_since_last = current_time - self._last_chunk_time
+                        if time_since_last < self.block_duration:
+                            time.sleep(self.block_duration - time_since_last)
+                        
+                        chunk_reshaped = chunk.reshape(1, -1)
+                        self.stream.on_next(chunk_reshaped)
+                        self._last_chunk_time = time.time()
+                        
+            except Empty:
+                with self._buffer_lock:
+                    if len(self._buffer) > 0 and time.time() - self._last_chunk_time > self.block_duration:
+                        padded_chunk = np.zeros(self.block_size, dtype=np.float32)
+                        padded_chunk[:len(self._buffer)] = self._buffer
+                        self._buffer = np.array([], dtype=np.float32)
+                        
+                        chunk_reshaped = padded_chunk.reshape(1, -1)
+                        self.stream.on_next(chunk_reshaped)
+                        self._last_chunk_time = time.time()
+            except Exception as e:
+                logger.error(f"Error in audio processing thread: {e}")
+                self.stream.on_error(e)
+                break
+        
+        with self._buffer_lock:
+            if len(self._buffer) > 0:
+                padded_chunk = np.zeros(self.block_size, dtype=np.float32)
+                padded_chunk[:len(self._buffer)] = self._buffer
+                chunk_reshaped = padded_chunk.reshape(1, -1)
+                self.stream.on_next(chunk_reshaped)
+        
+        self.stream.on_completed()
 
     def close(self):
         if not self._closed:
             self._closed = True
-            self.stream.on_completed()
             self._close_event.set()
 
     def push_audio(self, chunk: np.ndarray):
+        """Add audio chunk to the processing queue."""
         if not self._closed:
-            new_audio = np.expand_dims(chunk, axis=0)
+            if chunk.ndim > 1:
-            logger.debug('Add new chunk with shape:', new_audio.shape)
+                chunk = chunk.flatten()
-            self.stream.on_next(new_audio)
+            self._queue.put(chunk)
+            logger.debug(f'Added chunk to queue with {len(chunk)} samples')
 
 
 class DiartDiarization:
-    def __init__(self, sample_rate: int = 16000, config : SpeakerDiarizationConfig = None, use_microphone: bool = False):
+    def __init__(self, sample_rate: int = 16000, config : SpeakerDiarizationConfig = None, use_microphone: bool = False, block_duration: float = 0.5, segmentation_model_name: str = "pyannote/segmentation-3.0", embedding_model_name: str = "speechbrain/spkrec-ecapa-voxceleb"):
+        segmentation_model = m.SegmentationModel.from_pretrained(segmentation_model_name)
+        embedding_model = m.EmbeddingModel.from_pretrained(embedding_model_name)
+        
+        if config is None:
+            config = SpeakerDiarizationConfig(
+                segmentation=segmentation_model,
+                embedding=embedding_model,
+            )
+        
         self.pipeline = SpeakerDiarization(config=config)        
         self.observer = DiarizationObserver()
+        self.lag_diart = None
         
         if use_microphone:
-            self.source = MicrophoneAudioSource()
+            self.source = MicrophoneAudioSource(block_duration=block_duration)
             self.custom_source = None
         else:
-            self.custom_source = WebSocketAudioSource(uri="websocket_source", sample_rate=sample_rate)
+            self.custom_source = WebSocketAudioSource(
+                uri="websocket_source", 
+                sample_rate=sample_rate,
+                block_duration=block_duration
+            )
             self.source = self.custom_source
             
         self.inference = StreamingInference(
@@ -138,16 +214,102 @@ class DiartDiarization:
         if self.custom_source:
             self.custom_source.close()
 
-    def assign_speakers_to_tokens(self, end_attributed_speaker, tokens: list) -> float:
+    def assign_speakers_to_tokens(self, end_attributed_speaker, tokens: list, use_punctuation_split: bool = False) -> float:
         """
         Assign speakers to tokens based on timing overlap with speaker segments.
         Uses the segments collected by the observer.
+        
+        If use_punctuation_split is True, uses punctuation marks to refine speaker boundaries.
         """
         segments = self.observer.get_segments()
         
+        # Debug logging
+        logger.debug(f"assign_speakers_to_tokens called with {len(tokens)} tokens")
+        logger.debug(f"Available segments: {len(segments)}")
+        for i, seg in enumerate(segments[:5]):  # Show first 5 segments
+            logger.debug(f"  Segment {i}: {seg.speaker} [{seg.start:.2f}-{seg.end:.2f}]")
+        
+        if not self.lag_diart and segments and tokens:
+            self.lag_diart = segments[0].start - tokens[0].start
         for token in tokens:
             for segment in segments:
-                if not (segment.end <= token.start or segment.start >= token.end):
+                if not (segment.end <= token.start + self.lag_diart or segment.start >= token.end + self.lag_diart):
                     token.speaker = extract_number(segment.speaker) + 1
                     end_attributed_speaker = max(token.end, end_attributed_speaker)
-        return end_attributed_speaker
+        
+        if use_punctuation_split and len(tokens) > 1:
+            punctuation_marks = {'.', '!', '?'}
+            
+            print("Here are the tokens:", 
+                  [(t.text, t.start, t.end, t.speaker) for t in tokens[:10]])
+            
+            segment_map = []
+            for segment in segments:
+                speaker_num = extract_number(segment.speaker) + 1
+                segment_map.append((segment.start, segment.end, speaker_num))
+            segment_map.sort(key=lambda x: x[0])
+            
+            i = 0
+            while i < len(tokens):
+                current_token = tokens[i]
+                
+                is_sentence_end = False
+                if current_token.text and current_token.text.strip():
+                    text = current_token.text.strip()
+                    if text[-1] in punctuation_marks:
+                        is_sentence_end = True
+                        logger.debug(f"Token {i} ends sentence: '{current_token.text}' at {current_token.end:.2f}s")
+                
+                if is_sentence_end and current_token.speaker != -1:
+                    punctuation_time = current_token.end
+                    current_speaker = current_token.speaker
+                    
+                    j = i + 1
+                    next_sentence_tokens = []                    
+                    while j < len(tokens):
+                        next_token = tokens[j]
+                        next_sentence_tokens.append(j)
+                        
+                        # Check if this token ends the next sentence
+                        if next_token.text and next_token.text.strip():
+                            if next_token.text.strip()[-1] in punctuation_marks:
+                                break
+                        j += 1
+                    
+                    if next_sentence_tokens:
+                        speaker_times = {}
+                        
+                        for idx in next_sentence_tokens:
+                            token = tokens[idx]
+                            # Find which segments overlap with this token
+                            for seg_start, seg_end, seg_speaker in segment_map:
+                                if not (seg_end <= token.start or seg_start >= token.end):
+                                    # Calculate overlap duration
+                                    overlap_start = max(seg_start, token.start)
+                                    overlap_end = min(seg_end, token.end)
+                                    overlap_duration = overlap_end - overlap_start
+                                    
+                                    if seg_speaker not in speaker_times:
+                                        speaker_times[seg_speaker] = 0
+                                    speaker_times[seg_speaker] += overlap_duration
+                        
+                        if speaker_times:
+                            dominant_speaker = max(speaker_times.items(), key=lambda x: x[1])[0]
+                            
+                            if dominant_speaker != current_speaker:
+                                logger.debug(f"  Speaker change after punctuation: {current_speaker} → {dominant_speaker}")
+                                
+                                for idx in next_sentence_tokens:
+                                    if tokens[idx].speaker != dominant_speaker:
+                                        logger.debug(f"    Reassigning token {idx} ('{tokens[idx].text}') to Speaker {dominant_speaker}")
+                                        tokens[idx].speaker = dominant_speaker
+                                        end_attributed_speaker = max(tokens[idx].end, end_attributed_speaker)
+                            else:
+                                for idx in next_sentence_tokens:
+                                    if tokens[idx].speaker == -1:
+                                        tokens[idx].speaker = current_speaker
+                                        end_attributed_speaker = max(tokens[idx].end, end_attributed_speaker)
+                
+                i += 1         
+        
+        return end_attributed_speaker

whisperlivekit/parse_args.py

@@ -0,0 +1,162 @@
+
+from argparse import ArgumentParser
+
+def parse_args():
+    parser = ArgumentParser(description="Whisper FastAPI Online Server")
+    parser.add_argument(
+        "--host",
+        type=str,
+        default="localhost",
+        help="The host address to bind the server to.",
+    )
+    parser.add_argument(
+        "--port", type=int, default=8000, help="The port number to bind the server to."
+    )
+    parser.add_argument(
+        "--warmup-file",
+        type=str,
+        default=None,
+        dest="warmup_file",
+        help="""
+        The path to a speech audio wav file to warm up Whisper so that the very first chunk processing is fast.
+        If not set, uses https://github.com/ggerganov/whisper.cpp/raw/master/samples/jfk.wav.
+        If False, no warmup is performed.
+        """,
+    )
+
+    parser.add_argument(
+        "--confidence-validation",
+        action="store_true",
+        help="Accelerates validation of tokens using confidence scores. Transcription will be faster but punctuation might be less accurate.",
+    )
+
+    parser.add_argument(
+        "--diarization",
+        action="store_true",
+        default=False,
+        help="Enable speaker diarization.",
+    )
+
+    parser.add_argument(
+        "--punctuation-split",
+        action="store_true",
+        default=False,
+        help="Use punctuation marks from transcription to improve speaker boundary detection. Requires both transcription and diarization to be enabled.",
+    )
+
+    parser.add_argument(
+        "--segmentation-model",
+        type=str,
+        default="pyannote/segmentation-3.0",
+        help="Hugging Face model ID for pyannote.audio segmentation model.",
+    )
+
+    parser.add_argument(
+        "--embedding-model",
+        type=str,
+        default="pyannote/embedding",
+        help="Hugging Face model ID for pyannote.audio embedding model.",
+    )
+
+    parser.add_argument(
+        "--no-transcription",
+        action="store_true",
+        help="Disable transcription to only see live diarization results.",
+    )
+    
+    parser.add_argument(
+        "--min-chunk-size",
+        type=float,
+        default=0.5,
+        help="Minimum audio chunk size in seconds. It waits up to this time to do processing. If the processing takes shorter time, it waits, otherwise it processes the whole segment that was received by this time.",
+    )
+    
+    parser.add_argument(
+        "--model",
+        type=str,
+        default="tiny",
+        help="Name size of the Whisper model to use (default: tiny). Suggested values: tiny.en,tiny,base.en,base,small.en,small,medium.en,medium,large-v1,large-v2,large-v3,large,large-v3-turbo. The model is automatically downloaded from the model hub if not present in model cache dir.",
+    )
+    
+    parser.add_argument(
+        "--model_cache_dir",
+        type=str,
+        default=None,
+        help="Overriding the default model cache dir where models downloaded from the hub are saved",
+    )
+    parser.add_argument(
+        "--model_dir",
+        type=str,
+        default=None,
+        help="Dir where Whisper model.bin and other files are saved. This option overrides --model and --model_cache_dir parameter.",
+    )
+    parser.add_argument(
+        "--lan",
+        "--language",
+        type=str,
+        default="auto",
+        help="Source language code, e.g. en,de,cs, or 'auto' for language detection.",
+    )
+    parser.add_argument(
+        "--task",
+        type=str,
+        default="transcribe",
+        choices=["transcribe", "translate"],
+        help="Transcribe or translate.",
+    )
+    parser.add_argument(
+        "--backend",
+        type=str,
+        default="faster-whisper",
+        choices=["faster-whisper", "whisper_timestamped", "mlx-whisper", "openai-api"],
+        help="Load only this backend for Whisper processing.",
+    )
+    parser.add_argument(
+        "--vac",
+        action="store_true",
+        default=False,
+        help="Use VAC = voice activity controller. Recommended. Requires torch.",
+    )
+    parser.add_argument(
+        "--vac-chunk-size", type=float, default=0.04, help="VAC sample size in seconds."
+    )
+
+    parser.add_argument(
+        "--no-vad",
+        action="store_true",
+        help="Disable VAD (voice activity detection).",
+    )
+    
+    parser.add_argument(
+        "--buffer_trimming",
+        type=str,
+        default="segment",
+        choices=["sentence", "segment"],
+        help='Buffer trimming strategy -- trim completed sentences marked with punctuation mark and detected by sentence segmenter, or the completed segments returned by Whisper. Sentence segmenter must be installed for "sentence" option.',
+    )
+    parser.add_argument(
+        "--buffer_trimming_sec",
+        type=float,
+        default=15,
+        help="Buffer trimming length threshold in seconds. If buffer length is longer, trimming sentence/segment is triggered.",
+    )
+    parser.add_argument(
+        "-l",
+        "--log-level",
+        dest="log_level",
+        choices=["DEBUG", "INFO", "WARNING", "ERROR", "CRITICAL"],
+        help="Set the log level",
+        default="DEBUG",
+    )
+    parser.add_argument("--ssl-certfile", type=str, help="Path to the SSL certificate file.", default=None)
+    parser.add_argument("--ssl-keyfile", type=str, help="Path to the SSL private key file.", default=None)
+
+
+    args = parser.parse_args()
+    
+    args.transcription = not args.no_transcription
+    args.vad = not args.no_vad    
+    delattr(args, 'no_transcription')
+    delattr(args, 'no_vad')
+    
+    return args

whisperlivekit/timed_objects.py

@@ -26,4 +26,7 @@ class Transcript(TimedText):
 
 @dataclass
 class SpeakerSegment(TimedText):
+    """Represents a segment of audio attributed to a specific speaker.
+    No text nor probability is associated with this segment.
+    """
     pass

whisperlivekit/web/live_transcription.html

@@ -308,6 +308,7 @@
         let waveCtx = waveCanvas.getContext("2d");
         let animationFrame = null;
         let waitingForStop = false;
+        let lastReceivedData = null;
         waveCanvas.width = 60 * (window.devicePixelRatio || 1);
         waveCanvas.height = 30 * (window.devicePixelRatio || 1);
         waveCtx.scale(window.devicePixelRatio || 1, window.devicePixelRatio || 1);
@@ -357,18 +358,31 @@
 
                 websocket.onclose = () => {
                     if (userClosing) {
-                        if (!statusText.textContent.includes("Recording stopped. Processing final audio")) { // This is a bit of a hack. We should have a better way to handle this. eg. using a status code.
+                        if (waitingForStop) {
-                            statusText.textContent = "Finished processing audio! Ready to record again.";
+                            statusText.textContent = "Processing finalized or connection closed.";
+                            if (lastReceivedData) {
+                                renderLinesWithBuffer(
+                                    lastReceivedData.lines || [],
+                                    lastReceivedData.buffer_diarization || "",
+                                    lastReceivedData.buffer_transcription || "",
+                                    0, 0, true // isFinalizing = true
+                                );
+                            }
                         }
-                        waitingForStop = false;
+                        // If ready_to_stop was received, statusText is already "Finished processing..."
+                        // and waitingForStop is false.
                     } else {
-                        statusText.textContent =
+                        statusText.textContent = "Disconnected from the WebSocket server. (Check logs if model is loading.)";
-                            "Disconnected from the WebSocket server. (Check logs if model is loading.)";
                         if (isRecording) {
-                            stopRecording();
+                            stopRecording(); 
                         }
                     }
-                    userClosing = false;
+                    isRecording = false;  
+                    waitingForStop = false; 
+                    userClosing = false;  
+                    lastReceivedData = null;  
+                    websocket = null;    
+                    updateUI();  
                 };
 
                 websocket.onerror = () => {
@@ -382,31 +396,39 @@
                     
                     // Check for status messages
                     if (data.type === "ready_to_stop") {
-                        console.log("Ready to stop, closing WebSocket");
+                        console.log("Ready to stop received, finalizing display and closing WebSocket.");
-
-                        // signal that we are not waiting for stop anymore
                         waitingForStop = false;
-                        recordButton.disabled = false; // this should be elsewhere
-                        console.log("Record button enabled");
 
-                        //Now we can close the WebSocket
+                        if (lastReceivedData) {
-                        if (websocket) {
+                            renderLinesWithBuffer(
-                            websocket.close();
+                                lastReceivedData.lines || [],
-                            websocket = null;
+                                lastReceivedData.buffer_diarization || "",
+                                lastReceivedData.buffer_transcription || "",
+                                0, // No more lag
+                                0, // No more lag
+                                true // isFinalizing = true
+                            );
                         }
-
+                        statusText.textContent = "Finished processing audio! Ready to record again.";
-
+                        recordButton.disabled = false;
                         
+                        if (websocket) {
+                            websocket.close(); // will trigger onclose
+                            // websocket = null; // onclose handle setting websocket to null
+                        }
                         return;
                     }
                     
+                    lastReceivedData = data; 
+                    
                     // Handle normal transcription updates
                     const { 
                         lines = [], 
                         buffer_transcription = "", 
                         buffer_diarization = "",
                         remaining_time_transcription = 0,
-                        remaining_time_diarization = 0
+                        remaining_time_diarization = 0,
+                        status = "active_transcription"
                     } = data;
                     
                     renderLinesWithBuffer(
@@ -414,13 +436,20 @@
                         buffer_diarization, 
                         buffer_transcription, 
                         remaining_time_diarization,
-                        remaining_time_transcription
+                        remaining_time_transcription,
+                        false,
+                        status
                     );
                 };
             });
         }
 
-        function renderLinesWithBuffer(lines, buffer_diarization, buffer_transcription, remaining_time_diarization, remaining_time_transcription) {
+        function renderLinesWithBuffer(lines, buffer_diarization, buffer_transcription, remaining_time_diarization, remaining_time_transcription, isFinalizing = false, current_status = "active_transcription") {
+            if (current_status === "no_audio_detected") {
+                linesTranscriptDiv.innerHTML = "<p style='text-align: center; color: #666; margin-top: 20px;'><em>No audio detected...</em></p>";
+                return; 
+            }
+
             const linesHtml = lines.map((item, idx) => {
                 let timeInfo = "";
                 if (item.beg !== undefined && item.end !== undefined) {
@@ -430,30 +459,46 @@
                 let speakerLabel = "";
                 if (item.speaker === -2) {
                     speakerLabel = `<span class="silence">Silence<span id='timeInfo'>${timeInfo}</span></span>`;
-                } else if (item.speaker == 0) {
+                } else if (item.speaker == 0 && !isFinalizing) {
                     speakerLabel = `<span class='loading'><span class="spinner"></span><span id='timeInfo'>${remaining_time_diarization} second(s) of audio are undergoing diarization</span></span>`;
                 } else if (item.speaker == -1) {
-                    speakerLabel = `<span id="speaker"><span id='timeInfo'>${timeInfo}</span></span>`;
+                    speakerLabel = `<span id="speaker">Speaker 1<span id='timeInfo'>${timeInfo}</span></span>`;
-                } else if (item.speaker !== -1) {
+                } else if (item.speaker !== -1 && item.speaker !== 0) {
                     speakerLabel = `<span id="speaker">Speaker ${item.speaker}<span id='timeInfo'>${timeInfo}</span></span>`;
                 }
 
-                let textContent = item.text;
+
-                if (idx === lines.length - 1) {
+                let currentLineText = item.text || "";
-                    speakerLabel += `<span class="label_transcription"><span class="spinner"></span>Transcription lag <span id='timeInfo'>${remaining_time_transcription}s</span></span>`
+
-                }
+                if (idx === lines.length - 1) { 
-                if (idx === lines.length - 1 && buffer_diarization) {
+                    if (!isFinalizing) {
-                    speakerLabel += `<span class="label_diarization"><span class="spinner"></span>Diarization lag<span id='timeInfo'>${remaining_time_diarization}s</span></span>`
+                        if (remaining_time_transcription > 0) {
-                    textContent += `<span class="buffer_diarization">${buffer_diarization}</span>`;
+                             speakerLabel += `<span class="label_transcription"><span class="spinner"></span>Transcription lag <span id='timeInfo'>${remaining_time_transcription}s</span></span>`;
-                }
+                        }
-                if (idx === lines.length - 1) {
+                        if (buffer_diarization && remaining_time_diarization > 0) {
-                    textContent += `<span class="buffer_transcription">${buffer_transcription}</span>`;
+                             speakerLabel += `<span class="label_diarization"><span class="spinner"></span>Diarization lag<span id='timeInfo'>${remaining_time_diarization}s</span></span>`;
+                        }
+                    }
+
+                    if (buffer_diarization) {
+                        if (isFinalizing) {
+                            currentLineText += (currentLineText.length > 0 && buffer_diarization.trim().length > 0 ? " " : "") + buffer_diarization.trim();
+                        } else {
+                            currentLineText += `<span class="buffer_diarization">${buffer_diarization}</span>`;
+                        }
+                    }
+                    if (buffer_transcription) {
+                        if (isFinalizing) {
+                            currentLineText += (currentLineText.length > 0 && buffer_transcription.trim().length > 0 ? " " : "") + buffer_transcription.trim();
+                        } else {
+                            currentLineText += `<span class="buffer_transcription">${buffer_transcription}</span>`;
+                        }
+                    }
                 }
                 
-
+                return currentLineText.trim().length > 0 || speakerLabel.length > 0
-                return textContent
+                    ? `<p>${speakerLabel}<br/><div class='textcontent'>${currentLineText}</div></p>`
-                    ? `<p>${speakerLabel}<br/><div class='textcontent'>${textContent}</div></p>`
+                    : `<p>${speakerLabel}<br/></p>`; 
-                    : `<p>${speakerLabel}<br/></p>`;
             }).join("");
 
             linesTranscriptDiv.innerHTML = linesHtml;
@@ -578,20 +623,6 @@
             timerElement.textContent = "00:00";
             startTime = null;
             
-            if (websocket && websocket.readyState === WebSocket.OPEN) {
-                try {
-                    await websocket.send(JSON.stringify({
-                        type: "stop",
-                        message: "User stopped recording"
-                    }));
-                    statusText.textContent = "Recording stopped. Processing final audio...";
-                } catch (e) {
-                    console.error("Could not send stop message:", e);
-                    statusText.textContent = "Recording stopped. Error during final audio processing.";
-                    websocket.close();
-                    websocket = null;
-                }
-            }
             
             isRecording = false;
             updateUI();	
@@ -625,19 +656,22 @@
 
         function updateUI() {
             recordButton.classList.toggle("recording", isRecording);
-            
+            recordButton.disabled = waitingForStop;
+
             if (waitingForStop) {
-                statusText.textContent = "Please wait for processing to complete...";
+                if (statusText.textContent !== "Recording stopped. Processing final audio...") {
-                recordButton.disabled = true;  // Optionally disable the button while waiting
+                     statusText.textContent = "Please wait for processing to complete...";
-                console.log("Record button disabled");
+                }
             } else if (isRecording) {
                 statusText.textContent = "Recording...";
-                recordButton.disabled = false;
-                console.log("Record button enabled");
             } else {
-                statusText.textContent = "Click to start transcription";
+                if (statusText.textContent !== "Finished processing audio! Ready to record again." &&
+                    statusText.textContent !== "Processing finalized or connection closed.") {
+                    statusText.textContent = "Click to start transcription";
+                }
+            }
+            if (!waitingForStop) {
                 recordButton.disabled = false;
-                console.log("Record button enabled");
             }
         }
 
@@ -645,4 +679,4 @@
     </script>
 </body>
 
-</html>
+</html>

whisperlivekit/web/web_interface.py

@@ -0,0 +1,13 @@
+import logging
+import importlib.resources as resources
+
+logger = logging.getLogger(__name__)
+
+def get_web_interface_html():
+    """Loads the HTML for the web interface using importlib.resources."""
+    try:
+        with resources.files('whisperlivekit.web').joinpath('live_transcription.html').open('r', encoding='utf-8') as f:
+            return f.read()
+    except Exception as e:
+        logger.error(f"Error loading web interface HTML: {e}")
+        return "<html><body><h1>Error loading interface</h1></body></html>"

whisperlivekit/whisper_streaming_custom/online_asr.py

@@ -144,7 +144,11 @@ class OnlineASRProcessor:
         self.transcript_buffer.last_committed_time = self.buffer_time_offset
         self.committed: List[ASRToken] = []
 
-    def insert_audio_chunk(self, audio: np.ndarray):
+    def get_audio_buffer_end_time(self) -> float:
+        """Returns the absolute end time of the current audio_buffer."""
+        return self.buffer_time_offset + (len(self.audio_buffer) / self.SAMPLING_RATE)
+
+    def insert_audio_chunk(self, audio: np.ndarray, audio_stream_end_time: Optional[float] = None):
         """Append an audio chunk (a numpy array) to the current audio buffer."""
         self.audio_buffer = np.append(self.audio_buffer, audio)
 
@@ -179,18 +183,19 @@ class OnlineASRProcessor:
         return self.concatenate_tokens(self.transcript_buffer.buffer)
         
 
-    def process_iter(self) -> Transcript:
+    def process_iter(self) -> Tuple[List[ASRToken], float]:
         """
         Processes the current audio buffer.
 
-        Returns a Transcript object representing the committed transcript.
+        Returns a tuple: (list of committed ASRToken objects, float representing the audio processed up to time).
         """
+        current_audio_processed_upto = self.get_audio_buffer_end_time()
         prompt_text, _ = self.prompt()
         logger.debug(
             f"Transcribing {len(self.audio_buffer)/self.SAMPLING_RATE:.2f} seconds from {self.buffer_time_offset:.2f}"
         )
         res = self.asr.transcribe(self.audio_buffer, init_prompt=prompt_text)
-        tokens = self.asr.ts_words(res)  # Expecting List[ASRToken]
+        tokens = self.asr.ts_words(res)
         self.transcript_buffer.insert(tokens, self.buffer_time_offset)
         committed_tokens = self.transcript_buffer.flush()
         self.committed.extend(committed_tokens)
@@ -210,7 +215,7 @@ class OnlineASRProcessor:
         logger.debug(
             f"Length of audio buffer now: {len(self.audio_buffer)/self.SAMPLING_RATE:.2f} seconds"
         )
-        return committed_tokens
+        return committed_tokens, current_audio_processed_upto
 
     def chunk_completed_sentence(self):
         """
@@ -343,15 +348,17 @@ class OnlineASRProcessor:
                 )
                 sentences.append(sentence)
         return sentences
-    def finish(self) -> Transcript:
+    
+    def finish(self) -> Tuple[List[ASRToken], float]:
         """
         Flush the remaining transcript when processing ends.
+        Returns a tuple: (list of remaining ASRToken objects, float representing the final audio processed up to time).
         """
         remaining_tokens = self.transcript_buffer.buffer
-        final_transcript = self.concatenate_tokens(remaining_tokens)
+        logger.debug(f"Final non-committed tokens: {remaining_tokens}")
-        logger.debug(f"Final non-committed transcript: {final_transcript}")
+        final_processed_upto = self.buffer_time_offset + (len(self.audio_buffer) / self.SAMPLING_RATE)
-        self.buffer_time_offset += len(self.audio_buffer) / self.SAMPLING_RATE
+        self.buffer_time_offset = final_processed_upto
-        return final_transcript
+        return remaining_tokens, final_processed_upto
 
     def concatenate_tokens(
         self,
@@ -384,7 +391,8 @@ class VACOnlineASRProcessor:
     def __init__(self, online_chunk_size: float, *args, **kwargs):
         self.online_chunk_size = online_chunk_size
         self.online = OnlineASRProcessor(*args, **kwargs)
-
+        self.asr = self.online.asr
+        
         # Load a VAD model (e.g. Silero VAD)
         import torch
         model, _ = torch.hub.load(repo_or_dir="snakers4/silero-vad", model="silero_vad")
@@ -392,28 +400,35 @@ class VACOnlineASRProcessor:
 
         self.vac = FixedVADIterator(model)
         self.logfile = self.online.logfile
+        self.last_input_audio_stream_end_time: float = 0.0
         self.init()
 
     def init(self):
         self.online.init()
         self.vac.reset_states()
         self.current_online_chunk_buffer_size = 0
+        self.last_input_audio_stream_end_time = self.online.buffer_time_offset
         self.is_currently_final = False
         self.status: Optional[str] = None  # "voice" or "nonvoice"
         self.audio_buffer = np.array([], dtype=np.float32)
         self.buffer_offset = 0  # in frames
 
+    def get_audio_buffer_end_time(self) -> float:
+        """Returns the absolute end time of the audio processed by the underlying OnlineASRProcessor."""
+        return self.online.get_audio_buffer_end_time()
+
     def clear_buffer(self):
         self.buffer_offset += len(self.audio_buffer)
         self.audio_buffer = np.array([], dtype=np.float32)
 
-    def insert_audio_chunk(self, audio: np.ndarray):
+    def insert_audio_chunk(self, audio: np.ndarray, audio_stream_end_time: float):
         """
         Process an incoming small audio chunk:
           - run VAD on the chunk,
           - decide whether to send the audio to the online ASR processor immediately,
           - and/or to mark the current utterance as finished.
         """
+        self.last_input_audio_stream_end_time = audio_stream_end_time
         res = self.vac(audio)
         self.audio_buffer = np.append(self.audio_buffer, audio)
 
@@ -455,10 +470,11 @@ class VACOnlineASRProcessor:
                 self.buffer_offset += max(0, len(self.audio_buffer) - self.SAMPLING_RATE)
                 self.audio_buffer = self.audio_buffer[-self.SAMPLING_RATE:]
 
-    def process_iter(self) -> Transcript:
+    def process_iter(self) -> Tuple[List[ASRToken], float]:
         """
         Depending on the VAD status and the amount of accumulated audio,
         process the current audio chunk.
+        Returns a tuple: (list of committed ASRToken objects, float representing the audio processed up to time).
         """
         if self.is_currently_final:
             return self.finish()
@@ -467,17 +483,20 @@ class VACOnlineASRProcessor:
             return self.online.process_iter()
         else:
             logger.debug("No online update, only VAD")
-            return Transcript(None, None, "")
+            return [], self.last_input_audio_stream_end_time
 
-    def finish(self) -> Transcript:
+    def finish(self) -> Tuple[List[ASRToken], float]:
-        """Finish processing by flushing any remaining text."""
+        """
-        result = self.online.finish()
+        Finish processing by flushing any remaining text.
+        Returns a tuple: (list of remaining ASRToken objects, float representing the final audio processed up to time).
+        """
+        result_tokens, processed_upto = self.online.finish()
         self.current_online_chunk_buffer_size = 0
         self.is_currently_final = False
-        return result
+        return result_tokens, processed_upto
     
     def get_buffer(self):
         """
         Get the unvalidated buffer in string format.
         """
-        return self.online.concatenate_tokens(self.online.transcript_buffer.buffer).text
+        return self.online.concatenate_tokens(self.online.transcript_buffer.buffer)