Update app.py
Browse files
app.py
CHANGED
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@@ -14,14 +14,15 @@ MODEL_SIZES = {
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"Base (Faster)": "openai/whisper-base",
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"Small (Balanced)": "openai/whisper-small",
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"Distil-Large-v3 (General Purpose)": "distil-whisper/distil-large-v3",
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# FIX: The model 'distil-whisper/distil-large-v3-fr' does not exist.
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# We use the general distil-large-v3 and rely on the code below to force French.
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"Distil-Large-v3-FR (French-Specific)": "distil-whisper/distil-large-v3"
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}
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# Use a dictionary to cache loaded models
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model_cache = {}
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def get_model_pipeline(model_name, progress):
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"""
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Initializes and caches the ASR pipeline for a given model name.
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@@ -29,7 +30,6 @@ def get_model_pipeline(model_name, progress):
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if model_name not in model_cache:
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progress(0, desc="π Initializing ZeroGPU instance...")
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model_id = MODEL_SIZES[model_name]
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# Use GPU if available, otherwise fallback to CPU
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device = 0 if torch.cuda.is_available() else "cpu"
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progress(0.1, desc=f"β³ Loading {model_name} model...")
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@@ -37,12 +37,15 @@ def get_model_pipeline(model_name, progress):
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"automatic-speech-recognition",
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model=model_id,
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device=device,
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# Set max_new_tokens for generation, common for ASR
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max_new_tokens=128
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)
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progress(0.5, desc="β
Model loaded successfully!")
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return model_cache[model_name]
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def create_vtt(segments, file_path):
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"""
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Creates a WebVTT (.vtt) file from transcription segments.
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@@ -50,7 +53,6 @@ def create_vtt(segments, file_path):
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with open(file_path, "w", encoding="utf-8") as f:
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f.write("WEBVTT\n\n")
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for i, segment in enumerate(segments):
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# Calculate time strings in "HH:MM:SS.mmm" format
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start_ms = int(segment.get('start', 0) * 1000)
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end_ms = int(segment.get('end', 0) * 1000)
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@@ -77,9 +79,8 @@ def create_docx(segments, file_path, with_timestamps):
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if with_timestamps:
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for segment in segments:
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text = segment.get('text', '').strip()
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end = str(datetime.timedelta(seconds=int(segment.get('end', 0))))
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document.add_paragraph(f"[{start} - {end}] {text}")
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else:
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full_text = " ".join([segment.get('text', '').strip() for segment in segments])
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@@ -87,101 +88,123 @@ def create_docx(segments, file_path, with_timestamps):
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document.save(file_path)
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def
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"""
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Added logic for 5-minute sequencing.
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"""
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if audio_file is None:
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return (
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start_time = time.time()
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pipe = get_model_pipeline(model_size, progress)
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# Define generation arguments
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generate_kwargs = {}
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if model_size == "Distil-Large-v3-FR (French-Specific)":
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# Force French for this specific option
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generate_kwargs["language"] = "fr"
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full_text_list = []
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# --- New 5-Minute Sequencing Logic ---
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if sequence_5_min:
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progress(0.70, desc="βοΈ Splitting audio into 5-minute chunks...")
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audio = AudioSegment.from_file(audio_file)
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chunk_length_ms = 5 * 60 * 1000 # 5 minutes in milliseconds
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total_duration_ms = len(audio)
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num_chunks = (total_duration_ms +
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for i in range(num_chunks):
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start_ms = i *
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end_ms = min((i + 1) *
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chunk.export(temp_chunk_path, format="mp3")
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chunk_segments = chunk_output.get("chunks", [])
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for segment in chunk_segments:
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segment['start'] = segment.get('start', 0.0) + offset
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segment['end'] = segment.get('end', 0.0) + offset
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full_segments.append(segment)
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#
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if
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outputs = {}
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progress(0.85, desc="π Generating output files...")
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# Generate VTT
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if vtt_output:
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vtt_path = "transcription.vtt"
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create_vtt(full_segments, vtt_path)
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outputs["VTT"] = vtt_path
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# Generate DOCX with timestamps
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if docx_timestamp_output:
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docx_ts_path = "transcription_with_timestamps.docx"
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create_docx(full_segments, docx_ts_path, with_timestamps=True)
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outputs["DOCX (with timestamps)"] = docx_ts_path
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# Generate DOCX without timestamps
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if docx_no_timestamp_output:
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docx_no_ts_path = "transcription_without_timestamps.docx"
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create_docx(full_segments, docx_no_ts_path, with_timestamps=False)
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@@ -195,14 +218,14 @@ def transcribe_and_export(audio_file, model_size, vtt_output, docx_timestamp_out
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return (
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transcribed_text,
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gr.Files(value=downloadable_files, label="Download Transcripts"),
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gr.Audio(value=None),
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status_message
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)
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# --- Gradio UI ---
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with gr.Blocks(title="Whisper ZeroGPU Transcription") as demo:
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gr.Markdown("# ποΈ Whisper ZeroGPU Transcription")
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gr.Markdown("
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with gr.Row():
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audio_input = gr.Audio(sources=["microphone", "upload"], type="filepath", label="Audio File")
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@@ -213,13 +236,18 @@ with gr.Blocks(title="Whisper ZeroGPU Transcription") as demo:
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choices=list(MODEL_SIZES.keys()),
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value="Distil-Large-v3-FR (French-Specific)"
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)
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gr.Markdown("### Processing Options")
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# NEW CHECKBOX for 5-minute sequencing
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sequence_checkbox = gr.Checkbox(
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label="Process in 5-minute sequences (Recommended for files > 30 min or to prevent memory errors)",
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value=False
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)
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gr.Markdown("### Choose Output Formats")
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with gr.Row():
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vtt_checkbox = gr.Checkbox(label="VTT", value=True)
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@@ -232,10 +260,17 @@ with gr.Blocks(title="Whisper ZeroGPU Transcription") as demo:
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transcription_output = gr.Textbox(label="Full Transcription", lines=10)
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downloadable_files_output = gr.Files(label="Download Transcripts")
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transcribe_btn.click(
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fn=transcribe_and_export,
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inputs=[audio_input, model_selector, vtt_checkbox, docx_ts_checkbox, docx_no_ts_checkbox, sequence_checkbox],
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outputs=[transcription_output, downloadable_files_output, audio_input, status_text]
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)
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"Base (Faster)": "openai/whisper-base",
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"Small (Balanced)": "openai/whisper-small",
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"Distil-Large-v3 (General Purpose)": "distil-whisper/distil-large-v3",
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"Distil-Large-v3-FR (French-Specific)": "distil-whisper/distil-large-v3"
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}
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# Use a dictionary to cache loaded models
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model_cache = {}
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# Define the fixed chunk length (5 minutes in milliseconds)
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CHUNK_LENGTH_MS = 5 * 60 * 1000
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def get_model_pipeline(model_name, progress):
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"""
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Initializes and caches the ASR pipeline for a given model name.
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if model_name not in model_cache:
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progress(0, desc="π Initializing ZeroGPU instance...")
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model_id = MODEL_SIZES[model_name]
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device = 0 if torch.cuda.is_available() else "cpu"
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progress(0.1, desc=f"β³ Loading {model_name} model...")
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"automatic-speech-recognition",
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model=model_id,
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device=device,
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max_new_tokens=128
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)
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progress(0.5, desc="β
Model loaded successfully!")
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return model_cache[model_name]
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# Helper function to format seconds to HH:MM:SS string
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def format_seconds(seconds):
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return str(datetime.timedelta(seconds=int(seconds)))
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def create_vtt(segments, file_path):
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"""
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Creates a WebVTT (.vtt) file from transcription segments.
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with open(file_path, "w", encoding="utf-8") as f:
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f.write("WEBVTT\n\n")
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for i, segment in enumerate(segments):
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start_ms = int(segment.get('start', 0) * 1000)
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end_ms = int(segment.get('end', 0) * 1000)
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if with_timestamps:
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for segment in segments:
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text = segment.get('text', '').strip()
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start = format_seconds(segment.get('start', 0))
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end = format_seconds(segment.get('end', 0))
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document.add_paragraph(f"[{start} - {end}] {text}")
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else:
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full_text = " ".join([segment.get('text', '').strip() for segment in segments])
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document.save(file_path)
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# --- NEW FUNCTION: Analyze Audio and Populate Dropdown ---
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def analyze_audio_and_get_chunks(audio_file):
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"""
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Reads the audio file and generates chunk options for the dropdown.
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"""
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if audio_file is None:
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return gr.Dropdown(choices=["Full Audio"], value="Full Audio", interactive=False), "Please upload an audio file first."
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try:
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audio = AudioSegment.from_file(audio_file)
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total_duration_ms = len(audio)
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num_chunks = (total_duration_ms + CHUNK_LENGTH_MS - 1) // CHUNK_LENGTH_MS
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chunk_options = ["Full Audio"]
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for i in range(num_chunks):
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start_ms = i * CHUNK_LENGTH_MS
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end_ms = min((i + 1) * CHUNK_LENGTH_MS, total_duration_ms)
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start_sec = start_ms / 1000
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end_sec = end_ms / 1000
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start_time_str = format_seconds(start_sec).split('.')[0]
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end_time_str = format_seconds(end_sec).split('.')[0]
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option_name = f"Chunk {i+1} ({start_time_str} - {end_time_str})"
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chunk_options.append(option_name)
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status = f"Audio analyzed. Duration: {format_seconds(total_duration_ms/1000.0)}. Found {num_chunks} chunks."
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return gr.Dropdown(choices=chunk_options, value="Full Audio", interactive=True), status
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except Exception as e:
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error_msg = f"Error analyzing audio: {e}"
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return gr.Dropdown(choices=["Full Audio"], value="Full Audio", interactive=False), error_msg
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# --------------------------------------------------------
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@spaces.GPU
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def transcribe_and_export(audio_file, model_size, chunk_choice, vtt_output, docx_timestamp_output, docx_no_timestamp_output, progress=gr.Progress()):
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"""
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Main function to transcribe audio and export to selected formats.
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Modified to process a single selected chunk or the full audio.
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"""
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if audio_file is None:
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return (None, None, None, "Please upload an audio file.")
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start_time = time.time()
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pipe = get_model_pipeline(model_size, progress)
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# 1. Determine which segment to process
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audio_segment_to_process = audio_file
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offset = 0.0 # Time offset for segment timestamps
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if chunk_choice != "Full Audio":
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progress(0.70, desc="βοΈ Preparing audio segment...")
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try:
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# Parse chunk number from choice string (e.g., "Chunk 2 (5:00:00 - 10:00:00)")
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chunk_num = int(chunk_choice.split(' ')[1]) - 1
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full_audio = AudioSegment.from_file(audio_file)
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total_duration_ms = len(full_audio)
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start_ms = chunk_num * CHUNK_LENGTH_MS
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end_ms = min((chunk_num + 1) * CHUNK_LENGTH_MS, total_duration_ms)
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# Slice the audio
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chunk = full_audio[start_ms:end_ms]
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temp_chunk_path = "/tmp/selected_chunk.mp3"
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chunk.export(temp_chunk_path, format="mp3")
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audio_segment_to_process = temp_chunk_path
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offset = start_ms / 1000.0 # Offset is the start time of the chunk in seconds
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except Exception as e:
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return (None, None, None, f"Error preparing audio chunk: {e}")
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# 2. Define generation arguments (Language)
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generate_kwargs = {}
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if model_size == "Distil-Large-v3-FR (French-Specific)":
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generate_kwargs["language"] = "fr"
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# 3. Transcribe the segment
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progress(0.75, desc=f"π€ Transcribing {chunk_choice}...")
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raw_output = pipe(
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audio_segment_to_process,
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return_timestamps="word",
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generate_kwargs=generate_kwargs
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)
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# 4. Process and adjust segments
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full_segments = raw_output.get("chunks", [])
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transcribed_text = raw_output.get('text', '').strip()
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# Adjust timestamps if a chunk was processed
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if chunk_choice != "Full Audio":
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for segment in full_segments:
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# Add the offset to the segment start and end times
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segment['start'] = segment.get('start', 0.0) + offset
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segment['end'] = segment.get('end', 0.0) + offset
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# Clean up the temporary file
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if os.path.exists(audio_segment_to_process):
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os.remove(audio_segment_to_process)
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# 5. Generate output files
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outputs = {}
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progress(0.85, desc="π Generating output files...")
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|
|
| 198 |
if vtt_output:
|
| 199 |
vtt_path = "transcription.vtt"
|
| 200 |
create_vtt(full_segments, vtt_path)
|
| 201 |
outputs["VTT"] = vtt_path
|
| 202 |
|
|
|
|
| 203 |
if docx_timestamp_output:
|
| 204 |
docx_ts_path = "transcription_with_timestamps.docx"
|
| 205 |
create_docx(full_segments, docx_ts_path, with_timestamps=True)
|
| 206 |
outputs["DOCX (with timestamps)"] = docx_ts_path
|
| 207 |
|
|
|
|
| 208 |
if docx_no_timestamp_output:
|
| 209 |
docx_no_ts_path = "transcription_without_timestamps.docx"
|
| 210 |
create_docx(full_segments, docx_no_ts_path, with_timestamps=False)
|
|
|
|
| 218 |
return (
|
| 219 |
transcribed_text,
|
| 220 |
gr.Files(value=downloadable_files, label="Download Transcripts"),
|
| 221 |
+
gr.Audio(value=None),
|
| 222 |
status_message
|
| 223 |
)
|
| 224 |
|
| 225 |
# --- Gradio UI ---
|
| 226 |
with gr.Blocks(title="Whisper ZeroGPU Transcription") as demo:
|
| 227 |
gr.Markdown("# ποΈ Whisper ZeroGPU Transcription")
|
| 228 |
+
gr.Markdown("1. **Upload** an audio file. 2. Click **'Analyze Audio'** to load the 5-minute chunks. 3. Select a chunk or **'Full Audio'** and click **'Transcribe'**.")
|
| 229 |
|
| 230 |
with gr.Row():
|
| 231 |
audio_input = gr.Audio(sources=["microphone", "upload"], type="filepath", label="Audio File")
|
|
|
|
| 236 |
choices=list(MODEL_SIZES.keys()),
|
| 237 |
value="Distil-Large-v3-FR (French-Specific)"
|
| 238 |
)
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| 239 |
|
| 240 |
+
# NEW: Button to analyze audio and populate chunk options
|
| 241 |
+
analyze_btn = gr.Button("Analyze Audio π", variant="secondary")
|
| 242 |
+
|
| 243 |
+
# NEW: Dropdown for chunk selection
|
| 244 |
+
chunk_selector = gr.Dropdown(
|
| 245 |
+
label="Select Audio Segment (5-minute chunks)",
|
| 246 |
+
choices=["Full Audio"],
|
| 247 |
+
value="Full Audio",
|
| 248 |
+
interactive=False # Disabled until audio is uploaded and analyzed
|
| 249 |
+
)
|
| 250 |
+
|
| 251 |
gr.Markdown("### Choose Output Formats")
|
| 252 |
with gr.Row():
|
| 253 |
vtt_checkbox = gr.Checkbox(label="VTT", value=True)
|
|
|
|
| 260 |
transcription_output = gr.Textbox(label="Full Transcription", lines=10)
|
| 261 |
downloadable_files_output = gr.Files(label="Download Transcripts")
|
| 262 |
|
| 263 |
+
# NEW: Link the analyze button to the analysis function
|
| 264 |
+
analyze_btn.click(
|
| 265 |
+
fn=analyze_audio_and_get_chunks,
|
| 266 |
+
inputs=[audio_input],
|
| 267 |
+
outputs=[chunk_selector, status_text]
|
| 268 |
+
)
|
| 269 |
+
|
| 270 |
+
# UPDATED: Link the transcribe button to the transcription function
|
| 271 |
transcribe_btn.click(
|
| 272 |
fn=transcribe_and_export,
|
| 273 |
+
inputs=[audio_input, model_selector, chunk_selector, vtt_checkbox, docx_ts_checkbox, docx_no_ts_checkbox],
|
|
|
|
| 274 |
outputs=[transcription_output, downloadable_files_output, audio_input, status_text]
|
| 275 |
)
|
| 276 |
|