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Update app.py
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app.py
CHANGED
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@@ -28,12 +28,12 @@ class OptimizedWhisperApp:
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"openai/whisper-tiny",
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"openai/whisper-base",
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"openai/whisper-small",
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"openai/whisper-medium",
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"openai/whisper-large-v2",
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"openai/whisper-large-v3",
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"distil-whisper/distil-medium.en",
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"distil-whisper/distil-large-v2",
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"ilsp/whisper_greek_dialect_of_lesbos"
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]
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def create_pipe(self, model_name, use_flash_attention=True):
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@@ -54,13 +54,13 @@ class OptimizedWhisperApp:
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attn_implementation = "flash_attention_2"
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logger.info("Using Flash Attention 2")
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else:
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attn_implementation = "sdpa"
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if use_flash_attention and not FLASH_ATTN_AVAILABLE:
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logger.info("Flash Attention requested but not available, using SDPA")
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else:
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logger.info(f"Using {attn_implementation}")
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# Load model directly
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model = AutoModelForSpeechSeq2Seq.from_pretrained(
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model_name,
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torch_dtype=torch_dtype,
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@@ -74,7 +74,7 @@ class OptimizedWhisperApp:
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# Load processor
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processor = AutoProcessor.from_pretrained(model_name)
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# Create pipeline manually
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pipe = pipeline(
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"automatic-speech-recognition",
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model=model,
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@@ -133,7 +133,7 @@ class OptimizedWhisperApp:
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logger.info(f"Settings: {model_name}, {language}, {task}")
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logger.info(f"Chunk length: {chunk_length_s}s, Batch size: {batch_size}")
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# Prepare generation kwargs
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generate_kwargs = {}
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# Only set language if not auto-detection and model supports multilingual
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@@ -156,7 +156,7 @@ class OptimizedWhisperApp:
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generate_kwargs["task"] = task
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logger.info(f"Set task: {task}")
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# Transcribe
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logger.info("Starting transcription...")
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outputs = self.pipe(
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audio_file,
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@@ -227,7 +227,7 @@ class OptimizedWhisperApp:
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output += f"Device: {device}\n"
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output += f"Data type: {dtype}\n"
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output += "\n=== OPTIMIZATIONS ===\n"
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output += "• Direct model loading (not pipeline abstraction)\n"
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@@ -277,7 +277,7 @@ def create_interface():
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Uses the same optimizations as high-performing Whisper spaces:
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- Direct model loading for better control
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- Flash Attention 2 support
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- Optimized chunking and batching
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- Conservative parameter handling
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"""
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@@ -296,13 +296,7 @@ def create_interface():
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# Audio input
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audio_input = gr.Audio(
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label="🎵 Upload Audio File",
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type="filepath"
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waveform_options=gr.WaveformOptions(
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waveform_color="#01C6FF",
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waveform_progress_color="#0066B4",
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skip_length=2,
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show_controls=True,
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)
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)
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# Model selection
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@@ -415,7 +409,7 @@ def create_interface():
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**General recommendations:**
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- **Medium model** often provides the best balance
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- **30-second chunks** work well for most audio
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- **Flash Attention** speeds up processing significantly
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- **Automatic language detection** usually works well
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### ⚡ Performance Tips
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"openai/whisper-tiny",
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"openai/whisper-base",
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"openai/whisper-small",
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"openai/whisper-medium",
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"openai/whisper-large-v2",
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"openai/whisper-large-v3",
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"distil-whisper/distil-medium.en",
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"distil-whisper/distil-large-v2",
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"ilsp/whisper_greek_dialect_of_lesbos"
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]
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def create_pipe(self, model_name, use_flash_attention=True):
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attn_implementation = "flash_attention_2"
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logger.info("Using Flash Attention 2")
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else:
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attn_implementation = "sdpa"
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if use_flash_attention and not FLASH_ATTN_AVAILABLE:
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logger.info("Flash Attention requested but not available, using SDPA")
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else:
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logger.info(f"Using {attn_implementation}")
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# Load model directly
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model = AutoModelForSpeechSeq2Seq.from_pretrained(
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model_name,
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torch_dtype=torch_dtype,
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# Load processor
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processor = AutoProcessor.from_pretrained(model_name)
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# Create pipeline manually
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pipe = pipeline(
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"automatic-speech-recognition",
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model=model,
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logger.info(f"Settings: {model_name}, {language}, {task}")
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logger.info(f"Chunk length: {chunk_length_s}s, Batch size: {batch_size}")
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# Prepare generation kwargs
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generate_kwargs = {}
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# Only set language if not auto-detection and model supports multilingual
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generate_kwargs["task"] = task
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logger.info(f"Set task: {task}")
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# Transcribe
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logger.info("Starting transcription...")
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outputs = self.pipe(
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audio_file,
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output += f"Device: {device}\n"
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output += f"Data type: {dtype}\n"
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output += f"Flash Attention 2 available: {FLASH_ATTN_AVAILABLE and is_flash_attn_2_available()}\n"
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output += "\n=== OPTIMIZATIONS ===\n"
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output += "• Direct model loading (not pipeline abstraction)\n"
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Uses the same optimizations as high-performing Whisper spaces:
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- Direct model loading for better control
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+
- Flash Attention 2 support (when available)
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- Optimized chunking and batching
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- Conservative parameter handling
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"""
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# Audio input
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audio_input = gr.Audio(
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label="🎵 Upload Audio File",
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type="filepath"
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)
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# Model selection
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**General recommendations:**
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- **Medium model** often provides the best balance
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- **30-second chunks** work well for most audio
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+
- **Flash Attention** speeds up processing significantly (when available)
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- **Automatic language detection** usually works well
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### ⚡ Performance Tips
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