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Browse files- src/ai/voxtral_spaces_analyzer.py +33 -46
- src/ui/spaces_interface.py +6 -16
src/ai/voxtral_spaces_analyzer.py
CHANGED
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@@ -41,15 +41,20 @@ class VoxtralSpacesAnalyzer:
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Args:
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model_name (str): Name of the Voxtral model to use (pre-quantized)
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"""
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#
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model_mapping = {
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"Voxtral-Mini-3B-2507": "
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"Voxtral-Small-24B-2507": "
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}
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self.model_name = model_mapping.get(model_name, "
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self.current_model_key = model_name
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-
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self.gpu_manager = ZeroGPUManager()
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self.token_tracker = TokenTracker("Transformers-HF-Spaces")
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@@ -62,11 +67,11 @@ class VoxtralSpacesAnalyzer:
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def switch_model(self, model_name: str):
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"""Switch to a different model (will reload if different)."""
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model_mapping = {
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"Voxtral-Mini-3B-2507": "
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"Voxtral-Small-24B-2507": "
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}
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new_model_path = model_mapping.get(model_name, "
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if self.model_name != new_model_path:
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print(f"🔄 Switching to {model_name}")
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@@ -88,37 +93,17 @@ class VoxtralSpacesAnalyzer:
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dtype = self.gpu_manager.dtype
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print(f"🔄 Loading {self.current_model_key} on {device} with {dtype}...")
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# Load processor
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-
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device_str = "cuda" if device == "cuda" else "mps" if device == "mps" else "cpu"
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print(f"📦 Loading {self.current_model_key} pre-quantized model")
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self.
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device_map=device_str
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)
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except Exception as e:
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if "quantization config" in str(e).lower():
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print(f"⚠️ Quantization config issue, trying alternative loading...")
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# Alternative: load config first and modify it
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config = AutoConfig.from_pretrained(self.model_name)
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if hasattr(config, 'quantization_config'):
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config.quantization_config = None
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self.model = VoxtralForConditionalGeneration.from_pretrained(
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self.model_name,
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config=config,
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torch_dtype=dtype,
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low_cpu_mem_usage=True,
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device_map=device_str
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)
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else:
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raise e
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print(f"✅ {self.current_model_key} loaded successfully on {device}")
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@@ -138,13 +123,15 @@ class VoxtralSpacesAnalyzer:
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audio = AudioSegment.from_file(wav_path)
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return len(audio) / (1000 * 60)
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def _create_time_chunks(self, wav_path: str
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"""Create time-based chunks for processing."""
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total_duration = self._get_audio_duration(wav_path) * 60 # seconds
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# Use
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chunk_minutes =
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max_chunk_seconds = chunk_minutes * 60
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if total_duration <= max_chunk_seconds:
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return [(0, total_duration)]
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@@ -179,17 +166,16 @@ class VoxtralSpacesAnalyzer:
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wav_path: str,
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language: str = "french",
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selected_sections: list = None,
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chunk_duration_minutes: int = 15,
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reference_speakers_data: str = None
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) -> Dict[str, str]:
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"""
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Analyze audio by chunks using Voxtral with Zero GPU.
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Args:
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wav_path (str): Path to audio file
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language (str): Expected language
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selected_sections (list): Analysis sections to include
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chunk_duration_minutes (int): Chunk duration in minutes
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reference_speakers_data (str): Speaker diarization data
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Returns:
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@@ -203,9 +189,10 @@ class VoxtralSpacesAnalyzer:
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duration = self._get_audio_duration(wav_path)
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print(f"🎵 Audio duration: {duration:.1f} minutes")
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# Create chunks with
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chunks = self._create_time_chunks(wav_path
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-
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chunk_summaries = []
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Args:
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model_name (str): Name of the Voxtral model to use (pre-quantized)
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"""
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# Use original Mistral models for HF Spaces
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model_mapping = {
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"Voxtral-Mini-3B-2507": "mistralai/Voxtral-Mini-3B-2507",
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"Voxtral-Small-24B-2507": "mistralai/Voxtral-Small-24B-2507"
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}
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self.model_name = model_mapping.get(model_name, "mistralai/Voxtral-Mini-3B-2507")
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self.current_model_key = model_name
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+
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# Optimized chunk durations for Zero GPU (different per model)
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self.chunk_durations = {
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"Voxtral-Mini-3B-2507": 15, # 15 minutes for Mini
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"Voxtral-Small-24B-2507": 10 # 10 minutes for Small (larger model)
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}
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self.gpu_manager = ZeroGPUManager()
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self.token_tracker = TokenTracker("Transformers-HF-Spaces")
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def switch_model(self, model_name: str):
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"""Switch to a different model (will reload if different)."""
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model_mapping = {
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"Voxtral-Mini-3B-2507": "mistralai/Voxtral-Mini-3B-2507",
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"Voxtral-Small-24B-2507": "mistralai/Voxtral-Small-24B-2507"
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}
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new_model_path = model_mapping.get(model_name, "mistralai/Voxtral-Mini-3B-2507")
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if self.model_name != new_model_path:
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print(f"🔄 Switching to {model_name}")
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dtype = self.gpu_manager.dtype
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print(f"🔄 Loading {self.current_model_key} on {device} with {dtype}...")
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# Load processor and model following HuggingFace reference implementation
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print(f"📦 Loading {self.current_model_key} (original Mistral model)")
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self.processor = AutoProcessor.from_pretrained(self.model_name)
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# Use reference implementation from HuggingFace docs
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self.model = VoxtralForConditionalGeneration.from_pretrained(
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self.model_name,
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torch_dtype=dtype,
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device_map=device
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)
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print(f"✅ {self.current_model_key} loaded successfully on {device}")
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audio = AudioSegment.from_file(wav_path)
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return len(audio) / (1000 * 60)
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+
def _create_time_chunks(self, wav_path: str) -> List[Tuple[float, float]]:
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"""Create time-based chunks for processing with model-optimized durations."""
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total_duration = self._get_audio_duration(wav_path) * 60 # seconds
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# Use model-specific optimized chunk duration for Zero GPU
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chunk_minutes = self.chunk_durations.get(self.current_model_key, 15)
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max_chunk_seconds = chunk_minutes * 60
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print(f"🎯 Using {chunk_minutes}min chunks optimized for {self.current_model_key} on Zero GPU")
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if total_duration <= max_chunk_seconds:
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return [(0, total_duration)]
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wav_path: str,
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language: str = "french",
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selected_sections: list = None,
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reference_speakers_data: str = None
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) -> Dict[str, str]:
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"""
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Analyze audio by chunks using Voxtral with Zero GPU.
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+
Uses model-optimized chunk durations (15min for Mini, 10min for Small).
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Args:
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wav_path (str): Path to audio file
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language (str): Expected language
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selected_sections (list): Analysis sections to include
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reference_speakers_data (str): Speaker diarization data
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Returns:
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duration = self._get_audio_duration(wav_path)
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print(f"🎵 Audio duration: {duration:.1f} minutes")
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# Create chunks with model-optimized duration
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chunks = self._create_time_chunks(wav_path)
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chunk_minutes = self.chunk_durations.get(self.current_model_key, 15)
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print(f"📦 Splitting into {len(chunks)} chunks of {chunk_minutes}min")
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chunk_summaries = []
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src/ui/spaces_interface.py
CHANGED
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@@ -220,7 +220,7 @@ def handle_speaker_rename(new_name):
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@gpu_inference(duration=300)
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def handle_direct_transcription(
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audio_file, hf_token, language, transcription_mode, model_key,
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selected_sections, diarization_data, start_trim, end_trim
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):
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"""Gestion de l'analyse directe adaptée pour HF Spaces."""
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initialize_components()
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@@ -239,12 +239,11 @@ def handle_direct_transcription(
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if analyzer.current_model_key != model_name:
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analyzer.switch_model(model_name)
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# Lancer l'analyse
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results = analyzer.analyze_audio_chunks(
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wav_path=audio_file,
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language="auto",
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selected_sections=selected_sections,
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chunk_duration_minutes=int(chunk_duration),
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reference_speakers_data=diarization_data
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)
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@@ -463,15 +462,7 @@ def create_spaces_interface():
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with gr.Column(elem_classes="processing-section"):
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gr.Markdown(UILabels.MAIN_ANALYSIS_TITLE)
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gr.Markdown(UILabels.MAIN_ANALYSIS_DESCRIPTION)
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-
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# Contrôle taille des chunks
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chunk_duration_slider = gr.Slider(
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minimum=5,
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maximum=25,
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value=15,
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step=5,
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label=UILabels.CHUNK_DURATION_LABEL
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)
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# Configuration des sections de résumé
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gr.Markdown(UILabels.SUMMARY_SECTIONS_TITLE)
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@@ -543,7 +534,7 @@ def create_spaces_interface():
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# Gestion de l'analyse directe (adaptée pour Transformers uniquement)
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def handle_analysis_direct(
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audio_file, hf_token, language, local_model, start_trim, end_trim,
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s_resume, s_discussions, s_plan_action, s_decisions, s_prochaines_etapes,
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s_sujets_principaux, s_points_importants, s_questions, s_elements_suivi
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):
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@@ -566,10 +557,10 @@ def create_spaces_interface():
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selected_sections = [section_key for is_selected, section_key in sections_checkboxes if is_selected]
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# Appeler la fonction d'analyse directe
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_, summary = handle_direct_transcription(
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audio_file, hf_token, language, transcription_mode,
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model_key, selected_sections, current_diarization_context, start_trim, end_trim
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)
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return summary
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@@ -611,7 +602,6 @@ def create_spaces_interface():
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local_model_choice,
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start_trim_input,
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end_trim_input,
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chunk_duration_slider,
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section_resume_executif,
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section_discussions,
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section_plan_action,
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@gpu_inference(duration=300)
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def handle_direct_transcription(
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audio_file, hf_token, language, transcription_mode, model_key,
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+
selected_sections, diarization_data, start_trim, end_trim
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):
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"""Gestion de l'analyse directe adaptée pour HF Spaces."""
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initialize_components()
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if analyzer.current_model_key != model_name:
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analyzer.switch_model(model_name)
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+
# Lancer l'analyse (chunk duration automatique selon le modèle)
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results = analyzer.analyze_audio_chunks(
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wav_path=audio_file,
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language="auto",
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selected_sections=selected_sections,
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reference_speakers_data=diarization_data
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)
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with gr.Column(elem_classes="processing-section"):
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gr.Markdown(UILabels.MAIN_ANALYSIS_TITLE)
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gr.Markdown(UILabels.MAIN_ANALYSIS_DESCRIPTION)
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gr.Markdown("*Chunk duration is automatically optimized: 15min for Mini, 10min for Small (Zero GPU optimization)*")
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# Configuration des sections de résumé
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gr.Markdown(UILabels.SUMMARY_SECTIONS_TITLE)
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# Gestion de l'analyse directe (adaptée pour Transformers uniquement)
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def handle_analysis_direct(
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audio_file, hf_token, language, local_model, start_trim, end_trim,
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s_resume, s_discussions, s_plan_action, s_decisions, s_prochaines_etapes,
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s_sujets_principaux, s_points_importants, s_questions, s_elements_suivi
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):
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selected_sections = [section_key for is_selected, section_key in sections_checkboxes if is_selected]
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+
# Appeler la fonction d'analyse directe (chunk duration automatique)
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_, summary = handle_direct_transcription(
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audio_file, hf_token, language, transcription_mode,
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model_key, selected_sections, current_diarization_context, start_trim, end_trim
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)
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return summary
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local_model_choice,
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start_trim_input,
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end_trim_input,
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section_resume_executif,
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section_discussions,
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section_plan_action,
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