update import paths

diarization/diarization_online.py

@@ -0,0 +1,90 @@
+import asyncio
+import re
+import threading
+import numpy as np
+
+from diart import SpeakerDiarization
+from diart.inference import StreamingInference
+from diart.sources import AudioSource
+from timed_objects import SpeakerSegment
+
+def extract_number(s: str) -> int:
+    m = re.search(r'\d+', s)
+    return int(m.group()) if m else None
+
+
+class WebSocketAudioSource(AudioSource):
+    """
+    Custom AudioSource that blocks in read() until close() is called.
+    Use push_audio() to inject PCM chunks.
+    """
+    def __init__(self, uri: str = "websocket", sample_rate: int = 16000):
+        super().__init__(uri, sample_rate)
+        self._closed = False
+        self._close_event = threading.Event()
+
+    def read(self):
+        self._close_event.wait()
+
+    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):
+        if not self._closed:
+            self.stream.on_next(np.expand_dims(chunk, axis=0))
+
+
+class DiartDiarization:
+    def __init__(self, sample_rate: int):
+        self.processed_time = 0
+        self.segment_speakers = []
+        self.speakers_queue = asyncio.Queue()
+        self.pipeline = SpeakerDiarization()
+        self.source = WebSocketAudioSource(uri="websocket_source", sample_rate=sample_rate)
+        self.inference = StreamingInference(
+            pipeline=self.pipeline,
+            source=self.source,
+            do_plot=False,
+            show_progress=False,
+        )
+        # Attache la fonction hook et démarre l'inférence en arrière-plan.
+        self.inference.attach_hooks(self._diar_hook)
+        asyncio.get_event_loop().run_in_executor(None, self.inference)
+
+    def _diar_hook(self, result):
+        annotation, audio = result
+        if annotation._labels:
+            for speaker, label in annotation._labels.items():
+                start = label.segments_boundaries_[0]
+                end = label.segments_boundaries_[-1]
+                if end > self.processed_time:
+                    self.processed_time = end
+                asyncio.create_task(self.speakers_queue.put(SpeakerSegment(
+                    speaker=speaker,
+                    start=start,
+                    end=end,
+                )))
+        else:
+            dur = audio.extent.end
+            if dur > self.processed_time:
+                self.processed_time = dur
+
+    async def diarize(self, pcm_array: np.ndarray):
+        self.source.push_audio(pcm_array)
+        self.segment_speakers.clear()
+        while not self.speakers_queue.empty():
+            self.segment_speakers.append(await self.speakers_queue.get())
+
+    def close(self):
+        self.source.close()
+
+    def assign_speakers_to_tokens(self, end_attributed_speaker, tokens: list) -> list:
+        for token in tokens:
+            for segment in self.segment_speakers:
+                if not (segment.end <= token.start or segment.start >= token.end):
+                    token.speaker = extract_number(segment.speaker) + 1
+                    end_attributed_speaker = max(token.end, end_attributed_speaker)
+        return end_attributed_speaker

whisper_fastapi_online_server.py

@@ -10,8 +10,8 @@ from fastapi import FastAPI, WebSocket, WebSocketDisconnect
 from fastapi.responses import HTMLResponse
 from fastapi.middleware.cors import CORSMiddleware
 
-from src.whisper_streaming.whisper_online import backend_factory, online_factory, add_shared_args
+from whisper_streaming_custom.whisper_online import backend_factory, online_factory, add_shared_args
-from src.whisper_streaming.timed_objects import ASRToken
+from timed_objects import ASRToken
 
 import math
 import logging
@@ -49,7 +49,7 @@ parser.add_argument(
 parser.add_argument(
     "--diarization",
     type=bool,
-    default=False,
+    default=True,
     help="Whether to enable speaker diarization.",
 )
 
@@ -157,7 +157,7 @@ async def lifespan(app: FastAPI):
         asr, tokenizer = None, None
 
     if args.diarization:
-        from src.diarization.diarization_online import DiartDiarization
+        from diarization.diarization_online import DiartDiarization
         diarization = DiartDiarization(SAMPLE_RATE)
     else :
         diarization = None
@@ -174,7 +174,7 @@ app.add_middleware(
 
 
 # Load demo HTML for the root endpoint
-with open("src/web/live_transcription.html", "r", encoding="utf-8") as f:
+with open("web/live_transcription.html", "r", encoding="utf-8") as f:
     html = f.read()
 
 async def start_ffmpeg_decoder():
@@ -277,24 +277,18 @@ async def results_formatter(shared_state, websocket):
             
             # Process tokens to create response
             previous_speaker = -10
-            lines = [
+            lines = []
-            ]
             last_end_diarized = 0
             undiarized_text = []
             
             for token in tokens:
                 speaker = token.speaker
-                # Handle diarization differently if diarization is enabled
                 if args.diarization:
-                    # If token is not yet processed by diarization
                     if (speaker == -1 or speaker == 0) and token.end >= end_attributed_speaker:
-                        # Add this token's text to undiarized buffer instead of creating a new line
                         undiarized_text.append(token.text)
                         continue
-                    # If speaker isn't assigned yet but should be (based on timestamp)
                     elif (speaker == -1 or speaker == 0) and token.end < end_attributed_speaker:
                         speaker = previous_speaker
-                    # Track last diarized token end time
                     if speaker not in [-1, 0]:
                         last_end_diarized = max(token.end, last_end_diarized)
 
@@ -314,7 +308,6 @@ async def results_formatter(shared_state, websocket):
                     lines[-1]["end"] = format_time(token.end)
                     lines[-1]["diff"] = round(token.end - last_end_diarized, 2)
             
-            # Update buffer_diarization with undiarized text
             if undiarized_text:
                 combined_buffer_diarization = sep.join(undiarized_text)
                 if buffer_transcription:
@@ -322,7 +315,6 @@ async def results_formatter(shared_state, websocket):
                 await shared_state.update_diarization(end_attributed_speaker, combined_buffer_diarization)
                 buffer_diarization = combined_buffer_diarization
                 
-            # Prepare response object
             if lines:
                 response = {
                     "lines": lines, 
@@ -350,7 +342,6 @@ async def results_formatter(shared_state, websocket):
             response_content = ' '.join([str(line['speaker']) + ' ' + line["text"] for line in lines]) + ' | ' + buffer_transcription + ' | ' + buffer_diarization
             
             if response_content != shared_state.last_response_content:
-                # Only send if there's actual content to send
                 if lines or buffer_transcription or buffer_diarization:
                     await websocket.send_json(response)
                     shared_state.last_response_content = response_content