Bump version to 0.1.6

https://github.com/QuentinFuxa/WhisperLiveKit/issues/106
https://github.com/QuentinFuxa/WhisperLiveKit/issues/90
https://github.com/QuentinFuxa/WhisperLiveKit/issues/87
https://github.com/QuentinFuxa/WhisperLiveKit/issues/81
https://github.com/QuentinFuxa/WhisperLiveKit/issues/2

LICENSE

@@ -1,21 +1,33 @@
 MIT License
 
-Copyright (c) 2023 ÚFAL
+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
+Permission is hereby granted, free of charge, to any person obtaining a copy  
-of this software and associated documentation files (the "Software"), to deal
+of this software and associated documentation files (the "Software"), to deal  
-in the Software without restriction, including without limitation the rights
+in the Software without restriction, including without limitation the rights  
-to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+to use, copy, modify, merge, publish, distribute, sublicense, and/or sell  
-copies of the Software, and to permit persons to whom the Software is
+copies of the Software, and to permit persons to whom the Software is  
 furnished to do so, subject to the following conditions:
 
-The above copyright notice and this permission notice shall be included in all
+The above copyright notice and this permission notice shall be included in all  
 copies or substantial portions of the Software.
 
-THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR  
-IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,  
-FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE  
-AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER  
-LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,  
-OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE  
 SOFTWARE.
+
+---
+
+Third-party components included in this software:
+
+- **whisper_streaming** by ÚFAL – MIT License – https://github.com/ufal/whisper_streaming  
+- **silero-vad** by Snakers4 – MIT License – https://github.com/snakers4/silero-vad  
+- **Diart** by juanmc2005 – MIT License – https://github.com/juanmc2005/diart

README.md

@@ -15,16 +15,16 @@
 
 ## 🚀 Overview
 
-This project is based on [Whisper Streaming](https://github.com/ufal/whisper_streaming) and lets you transcribe audio directly from your browser. WhisperLiveKit provides a complete backend solution for real-time speech transcription with an example frontend that you can customize for your own needs. Everything runs locally on your machine ✨
+This project is based on [Whisper Streaming](https://github.com/ufal/whisper_streaming) and lets you transcribe audio directly from your browser. WhisperLiveKit provides a complete backend solution for real-time speech transcription with a functional and simple frontend that you can customize for your own needs. Everything runs locally on your machine ✨
 
 ### 🔄 Architecture
 
-WhisperLiveKit consists of two main components:
+WhisperLiveKit consists of three main components:
 
-- **Backend (Server)**: FastAPI WebSocket server that processes audio and provides real-time transcription
+- **Frontend**: A basic HTML & JavaScript interface that captures microphone audio and streams it to the backend via WebSockets. You can use and adapt the provided template at [whisperlivekit/web/live_transcription.html](https://github.com/QuentinFuxa/WhisperLiveKit/blob/main/whisperlivekit/web/live_transcription.html) for your specific use case.
-- **Frontend Example**: Basic HTML & JavaScript implementation that demonstrates how to capture and stream audio
+- **Backend (Web Server)**: A FastAPI-based WebSocket server that receives streamed audio data, processes it in real time, and returns transcriptions to the frontend. This is where the WebSocket logic and routing live.
+- **Core Backend (Library Logic)**: A server-agnostic core that handles audio processing, ASR, and diarization. It exposes reusable components that take in audio bytes and return transcriptions. This makes it easy to plug into any WebSocket or audio stream pipeline.
 
-> **Note**: We recommend installing this library on the server/backend. For the frontend, you can use and adapt the provided HTML template from [whisperlivekit/web/live_transcription.html](https://github.com/QuentinFuxa/WhisperLiveKit/blob/main/whisperlivekit/web/live_transcription.html) for your specific use case.
 
 ### ✨ Key Features
 
@@ -33,13 +33,13 @@ WhisperLiveKit consists of two main components:
 - **🔒 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
  
-### ⚙️ Differences from [Whisper Streaming](https://github.com/ufal/whisper_streaming)
+### ⚙️ Core differences from [Whisper Streaming](https://github.com/ufal/whisper_streaming)
 
+- **Automatic Silence Chunking** – Automatically chunks when no audio is detected to limit buffer size
 - **Multi-User Support** – Handles multiple users simultaneously by decoupling backend and online ASR
+- **Confidence Validation** – Immediately validate high-confidence tokens for faster inference
 - **MLX Whisper Backend** – Optimized for Apple Silicon for faster local processing
 - **Buffering Preview** – Displays unvalidated transcription segments
-- **Confidence Validation** – Immediately validate high-confidence tokens for faster inference
-- **Apple Silicon Optimized** - MLX backend for faster local processing on Mac
 
 ## 📖 Quick Start
 
@@ -53,6 +53,14 @@ whisperlivekit-server --model tiny.en
 # Open your browser at http://localhost:8000
 ```
 
+### Quick Start with SSL
+```bash
+# You must provide a certificate and key
+whisperlivekit-server -ssl-certfile public.crt --ssl-keyfile private.key
+
+# Open your browser at https://localhost:8000
+```
+
 That's it! Start speaking and watch your words appear on screen.
 
 ## 🛠️ Installation Options
@@ -201,6 +209,8 @@ WhisperLiveKit offers extensive configuration options:
 | `--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` |
+| `--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` |
 
 ## 🔧 How It Works
 

setup.py

@@ -1,7 +1,7 @@
 from setuptools import setup, find_packages
 setup(
     name="whisperlivekit",
-    version="0.1.4",
+    version="0.1.6",
     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",

whisper_fastapi_online_server.py

@@ -1,82 +0,0 @@
-from contextlib import asynccontextmanager
-from fastapi import FastAPI, WebSocket, WebSocketDisconnect
-from fastapi.responses import HTMLResponse
-from fastapi.middleware.cors import CORSMiddleware
-
-from whisperlivekit import WhisperLiveKit
-from whisperlivekit.audio_processor import AudioProcessor
-
-import asyncio
-import logging
-import os
-
-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)
-
-kit = None
-
-@asynccontextmanager
-async def lifespan(app: FastAPI):
-    global kit
-    kit = WhisperLiveKit()
-    yield
-
-app = FastAPI(lifespan=lifespan)
-app.add_middleware(
-    CORSMiddleware,
-    allow_origins=["*"],
-    allow_credentials=True,
-    allow_methods=["*"],
-    allow_headers=["*"],
-)
-
-
-@app.get("/")
-async def get():
-    return HTMLResponse(kit.web_interface())
-
-
-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)
-    except Exception as e:
-        logger.warning(f"Error in WebSocket results handler: {e}")
-
-
-@app.websocket("/asr")
-async def websocket_endpoint(websocket: WebSocket):
-    audio_processor = AudioProcessor()
-
-    await websocket.accept()
-    logger.info("WebSocket connection opened.")
-            
-    results_generator = await audio_processor.create_tasks()
-    websocket_task = asyncio.create_task(handle_websocket_results(websocket, results_generator))
-
-    try:
-        while True:
-            message = await websocket.receive_bytes()
-            await audio_processor.process_audio(message)
-    except WebSocketDisconnect:
-        logger.warning("WebSocket disconnected.")
-    finally:
-        websocket_task.cancel()
-        await audio_processor.cleanup()
-        logger.info("WebSocket endpoint cleaned up.")
-
-if __name__ == "__main__":
-    import uvicorn
-    
-    temp_kit = WhisperLiveKit(transcription=False, diarization=False)
-    
-    uvicorn.run(
-        "whisper_fastapi_online_server:app", 
-        host=temp_kit.args.host, 
-        port=temp_kit.args.port, 
-        reload=True,
-        log_level="info"
-    )

whisperlivekit/audio_processor.py

@@ -15,6 +15,8 @@ logging.basicConfig(level=logging.INFO, format="%(asctime)s - %(levelname)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."""
     return str(timedelta(seconds=int(seconds)))
@@ -41,8 +43,9 @@ 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 = ""
@@ -62,6 +65,13 @@ class AudioProcessor:
         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()
+
+        # Task references
+        self.transcription_task = None
+        self.diarization_task = None
+        self.ffmpeg_reader_task = None
+        self.watchdog_task = None
+        self.all_tasks_for_cleanup = []
         
         # Initialize transcription engine if enabled
         if self.args.transcription:
@@ -205,24 +215,12 @@ class AudioProcessor:
                     self.last_ffmpeg_activity = time()
                     continue
 
-                # Reduce timeout for reading from FFmpeg
+                chunk = await loop.run_in_executor(None, self.ffmpeg_process.stdout.read, buffer_size)
-                try:
+                if chunk:
-                    chunk = await asyncio.wait_for(
-                        loop.run_in_executor(None, self.ffmpeg_process.stdout.read, buffer_size),
-                        timeout=5.0  # Shorter timeout (5 seconds instead of 15)
-                    )
-                    if chunk:
-                        self.last_ffmpeg_activity = time()
-                        
-                except asyncio.TimeoutError:
-                    logger.warning("FFmpeg read timeout. Restarting...")
-                    await self.restart_ffmpeg()
-                    beg = time()
                     self.last_ffmpeg_activity = time()
-                    continue
+                        
-
                 if not chunk:
-                    logger.info("FFmpeg stdout closed.")
+                    logger.info("FFmpeg stdout closed, no more data to read.")
                     break
                     
                 self.pcm_buffer.extend(chunk)
@@ -233,7 +231,7 @@ class AudioProcessor:
                         self.convert_pcm_to_float(self.pcm_buffer).copy()
                     )
 
-                # Process when we have enough data
+                # Process when enough data
                 if len(self.pcm_buffer) >= self.bytes_per_sec:
                     if len(self.pcm_buffer) > self.max_bytes_per_sec:
                         logger.warning(
@@ -257,6 +255,15 @@ 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."""
@@ -266,8 +273,23 @@ class AudioProcessor:
         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)
@@ -290,12 +312,15 @@ class AudioProcessor:
                 await self.update_transcription(
                     new_tokens, buffer, 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."""
@@ -304,6 +329,10 @@ 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)
@@ -315,12 +344,15 @@ class AudioProcessor:
                 )
                 
                 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."""
@@ -410,6 +442,19 @@ class AudioProcessor:
                     yield response
                     self.last_response_content = response_content
                 
+                # 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
                 
             except Exception as e:
@@ -419,65 +464,117 @@ 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)))
+            self.diarization_task = asyncio.create_task(self.diarization_processor(self.diarization))
+            self.all_tasks_for_cleanup.append(self.diarization_task)
+            processing_tasks_for_watchdog.append(self.diarization_task)
         
-        tasks.append(asyncio.create_task(self.ffmpeg_stdout_reader()))
+        self.ffmpeg_reader_task = asyncio.create_task(self.ffmpeg_stdout_reader())
-        
+        self.all_tasks_for_cleanup.append(self.ffmpeg_reader_task)
-        # Monitor overall system health
+        processing_tasks_for_watchdog.append(self.ffmpeg_reader_task)
-        async def watchdog():
-            while True:
-                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}")
-                    
-                    # Check for FFmpeg process health with shorter thresholds
-                    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()
-                            
-                except Exception as e:
-                    logger.error(f"Error in watchdog task: {e}")
 
-        tasks.append(asyncio.create_task(watchdog()))
+        # Monitor overall system health
-        self.tasks = tasks
+        self.watchdog_task = asyncio.create_task(self.watchdog(processing_tasks_for_watchdog))
+        self.all_tasks_for_cleanup.append(self.watchdog_task)
         
         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():
+                task.cancel()
+        
+        created_tasks = [t for t in self.all_tasks_for_cleanup if t]
+        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()
+                except Exception as e:
+                    logger.warning(f"Error closing ffmpeg stdin during cleanup: {e}")
             
-        try:
+            # Wait for ffmpeg process to terminate
-            await asyncio.gather(*self.tasks, return_exceptions=True)
+            if self.ffmpeg_process.poll() is None: # Check if process is still running
-            self.ffmpeg_process.stdin.close()
+                logger.info("Waiting for FFmpeg process to terminate...")
-            self.ffmpeg_process.wait()
+                try:
-        except Exception as e:
+                    # Run wait in executor to avoid blocking async loop
-            logger.warning(f"Error during cleanup: {e}")
+                    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
-        if self.args.diarization and hasattr(self, 'diarization'):
+                    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
         
@@ -492,20 +589,41 @@ class AudioProcessor:
                 if not self.ffmpeg_process or not hasattr(self.ffmpeg_process, 'stdin') or self.ffmpeg_process.poll() is not None:
                     logger.warning("FFmpeg process not available, restarting...")
                     await self.restart_ffmpeg()
-                    
-                self.ffmpeg_process.stdin.write(message)
-                self.ffmpeg_process.stdin.flush()
-                self.last_ffmpeg_activity = time()  # Update activity timestamp
-                return
                 
+                loop = asyncio.get_running_loop()                
+                try:
+                    await asyncio.wait_for(
+                        loop.run_in_executor(None, lambda: self.ffmpeg_process.stdin.write(message)),
+                        timeout=2.0
+                    )
+                except asyncio.TimeoutError:
+                    logger.warning("FFmpeg write operation timed out, restarting...")
+                    await self.restart_ffmpeg()
+                    retry_count += 1
+                    continue
+                    
+                try:
+                    await asyncio.wait_for(
+                        loop.run_in_executor(None, self.ffmpeg_process.stdin.flush),
+                        timeout=2.0
+                    )
+                except asyncio.TimeoutError:
+                    logger.warning("FFmpeg flush operation timed out, restarting...")
+                    await self.restart_ffmpeg()
+                    retry_count += 1
+                    continue
+                    
+                self.last_ffmpeg_activity = time()
+                return
+                    
             except (BrokenPipeError, AttributeError, OSError) as e:
                 retry_count += 1
                 logger.warning(f"Error writing to FFmpeg: {e}. Retry {retry_count}/{max_retries}...")
                 
                 if retry_count < max_retries:
                     await self.restart_ffmpeg()
-                    await asyncio.sleep(0.5)  # Shorter pause between retries
+                    await asyncio.sleep(0.5)
                 else:
                     logger.error("Maximum retries reached for FFmpeg process")
                     await self.restart_ffmpeg()
-                    return
+                    return

whisperlivekit/basic_server.py

@@ -3,12 +3,13 @@ from fastapi import FastAPI, WebSocket, WebSocketDisconnect
 from fastapi.responses import HTMLResponse
 from fastapi.middleware.cors import CORSMiddleware
 
-from whisperlivekit import WhisperLiveKit
+from whisperlivekit import WhisperLiveKit, parse_args
 from whisperlivekit.audio_processor import AudioProcessor
 
 import asyncio
 import logging
-import os
+import os, sys
+import argparse
 
 logging.basicConfig(level=logging.INFO, format="%(asctime)s - %(levelname)s - %(message)s")
 logging.getLogger().setLevel(logging.WARNING)
@@ -43,6 +44,11 @@ async def handle_websocket_results(websocket, results_generator):
     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}")
 
@@ -61,26 +67,58 @@ async def websocket_endpoint(websocket: WebSocket):
         while True:
             message = await websocket.receive_bytes()
             await audio_processor.process_audio(message)
+    except KeyError as e:
+        if 'bytes' in str(e):
+            logger.warning(f"Client has closed the connection.")
+        else:
+            logger.error(f"Unexpected KeyError in websocket_endpoint: {e}", exc_info=True)
     except WebSocketDisconnect:
-        logger.warning("WebSocket disconnected.")
+        logger.info("WebSocket disconnected by client during message receiving loop.")
+    except Exception as e:
+        logger.error(f"Unexpected error in websocket_endpoint main loop: {e}", exc_info=True)
     finally:
-        websocket_task.cancel()
+        logger.info("Cleaning up WebSocket endpoint...")
+        if not websocket_task.done():
+            websocket_task.cancel()
+        try:
+            await websocket_task
+        except asyncio.CancelledError:
+            logger.info("WebSocket results handler task was cancelled.")
+        except Exception as e:
+            logger.warning(f"Exception while awaiting websocket_task completion: {e}")
+            
         await audio_processor.cleanup()
-        logger.info("WebSocket endpoint cleaned up.")
+        logger.info("WebSocket endpoint cleaned up successfully.")
 
 def main():
     """Entry point for the CLI command."""
     import uvicorn
     
-    temp_kit = WhisperLiveKit(transcription=False, diarization=False)
+    args = parse_args()
     
-    uvicorn.run(
+    uvicorn_kwargs = {
-        "whisperlivekit.basic_server:app", 
+        "app": "whisperlivekit.basic_server:app",
-        host=temp_kit.args.host, 
+        "host":args.host, 
-        port=temp_kit.args.port, 
+        "port":args.port, 
-        reload=True,
+        "reload": False,
-        log_level="info"
+        "log_level": "info",
-    )
+        "lifespan": "on",
+    }
+    
+    ssl_kwargs = {}
+    if args.ssl_certfile or args.ssl_keyfile:
+        if not (args.ssl_certfile and args.ssl_keyfile):
+            raise ValueError("Both --ssl-certfile and --ssl-keyfile must be specified together.")
+        ssl_kwargs = {
+            "ssl_certfile": args.ssl_certfile,
+            "ssl_keyfile": args.ssl_keyfile
+        }
+
+
+    if ssl_kwargs:
+        uvicorn_kwargs = {**uvicorn_kwargs, **ssl_kwargs}
+
+    uvicorn.run(**uvicorn_kwargs)
 
 if __name__ == "__main__":
     main()

whisperlivekit/core.py

@@ -130,6 +130,9 @@ def parse_args():
         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()
     

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);
@@ -321,7 +322,8 @@
 
         const host = window.location.hostname || "localhost";
         const port = window.location.port || "8000";
-        const defaultWebSocketUrl = `ws://${host}:${port}/asr`;
+        const protocol = window.location.protocol === "https:" ? "wss" : "ws";
+        const defaultWebSocketUrl = `${protocol}://${host}:${port}/asr`;
         websocketInput.value = defaultWebSocketUrl;
         websocketUrl = defaultWebSocketUrl;
 
@@ -356,18 +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 = () => {
@@ -381,24 +396,31 @@
                     
                     // 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 = [], 
@@ -413,13 +435,14 @@
                         buffer_diarization, 
                         buffer_transcription, 
                         remaining_time_diarization,
-                        remaining_time_transcription
+                        remaining_time_transcription,
+                        false // isFinalizing = false for normal updates
                     );
                 };
             });
         }
 
-        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) {
             const linesHtml = lines.map((item, idx) => {
                 let timeInfo = "";
                 if (item.beg !== undefined && item.end !== undefined) {
@@ -429,30 +452,46 @@
                 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;
@@ -577,20 +616,6 @@
             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();	
@@ -624,19 +649,22 @@
 
         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");
             }
         }
 
@@ -644,4 +672,4 @@
     </script>
 </body>
 
-</html>
+</html>

whisperlivekit/whisper_streaming_custom/whisper_online.py

@@ -179,7 +179,7 @@ def warmup_asr(asr, warmup_file=None, timeout=5):
         logger.warning(f"Warmup file {warmup_file} invalid or missing.")
         return False
     
-    print(f"Warmping up Whisper with {warmup_file}")
+    print(f"Warming up Whisper with {warmup_file}")
     try:
         import librosa
         audio, sr = librosa.load(warmup_file, sr=16000)