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Add Qwen-ASR for automatic transcription, both sizes and keep Whisper as an option in dropdown.
#5
by Impulse2000 - opened
- app.py +50 -19
- requirements.txt +1 -0
app.py
CHANGED
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@@ -84,29 +84,55 @@ def decode_codes_to_audio(merged_codes):
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return audio[0, 0]
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@spaces.GPU(duration=60)
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def transcribe_audio(audio_path):
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if audio_path is None:
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raise gr.Error("Please upload a reference audio file first.")
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try:
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gr.Info("Transcribing audio with
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if not text:
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raise gr.Error("
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gr.Info(f"Detected language: {
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return text
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except gr.Error:
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raise
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@@ -240,7 +266,12 @@ with gr.Blocks(title="Fish Audio S2 Pro") as app:
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"The model will clone that voice for synthesis. Language is inferred automatically."
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)
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ref_audio = gr.Audio(label="Reference Audio", type="filepath")
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ref_text = gr.Textbox(
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label="Reference Audio Transcription",
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placeholder="Exact transcription of the reference audio, or click Auto-transcribe above...",
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@@ -314,7 +345,7 @@ with gr.Blocks(title="Fish Audio S2 Pro") as app:
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transcribe_btn.click(
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fn=transcribe_audio,
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inputs=[ref_audio],
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outputs=[ref_text],
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)
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@@ -325,4 +356,4 @@ with gr.Blocks(title="Fish Audio S2 Pro") as app:
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)
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if __name__ == "__main__":
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app.launch()
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return audio[0, 0]
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ASR_MODELS = {
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"Qwen3-ASR-1.7B β larger, more accurate": ("qwen", "Qwen/Qwen3-ASR-1.7B"),
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"Qwen3-ASR-0.6B β smaller, faster": ("qwen", "Qwen/Qwen3-ASR-0.6B"),
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"Whisper large-v3 (faster-whisper)": ("whisper", "large-v3"),
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}
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DEFAULT_ASR = "Qwen3-ASR-1.7B β larger, more accurate"
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asr_models = {}
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def get_asr_model(label):
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if label not in asr_models:
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backend, model_id = ASR_MODELS[label]
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if backend == "qwen":
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from qwen_asr import Qwen3ASRModel
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asr_models[label] = Qwen3ASRModel.from_pretrained(
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model_id,
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dtype=torch.bfloat16,
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device_map="cuda:0" if torch.cuda.is_available() else "cpu",
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max_inference_batch_size=32,
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max_new_tokens=256,
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)
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else:
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from faster_whisper import WhisperModel
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device = "cuda" if torch.cuda.is_available() else "cpu"
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asr_models[label] = WhisperModel(model_id, device=device, compute_type="int8")
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return asr_models[label]
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@spaces.GPU(duration=60)
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def transcribe_audio(audio_path, asr_label):
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if audio_path is None:
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raise gr.Error("Please upload a reference audio file first.")
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try:
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gr.Info(f"Transcribing audio with {asr_label}...")
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backend, _ = ASR_MODELS[asr_label]
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model = get_asr_model(asr_label)
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if backend == "qwen":
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result = model.transcribe(audio=audio_path, language=None)[0]
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text = (result.text or "").strip()
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detected_language = result.language
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else:
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segments, info = model.transcribe(audio_path, beam_size=5, vad_filter=True)
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text = " ".join(seg.text.strip() for seg in segments).strip()
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detected_language = info.language
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if not text:
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raise gr.Error("No speech could be detected in the audio.")
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gr.Info(f"Detected language: {detected_language}")
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return text
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except gr.Error:
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raise
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"The model will clone that voice for synthesis. Language is inferred automatically."
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)
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ref_audio = gr.Audio(label="Reference Audio", type="filepath")
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asr_model_selector = gr.Radio(
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choices=list(ASR_MODELS.keys()),
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value=DEFAULT_ASR,
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label="ASR Model",
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)
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transcribe_btn = gr.Button("π€ Auto-transcribe", variant="secondary", size="sm")
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ref_text = gr.Textbox(
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label="Reference Audio Transcription",
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placeholder="Exact transcription of the reference audio, or click Auto-transcribe above...",
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transcribe_btn.click(
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fn=transcribe_audio,
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inputs=[ref_audio, asr_model_selector],
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outputs=[ref_text],
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)
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)
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if __name__ == "__main__":
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app.launch(server_port=8181)
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requirements.txt
CHANGED
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@@ -6,6 +6,7 @@ datasets
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lightning
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hydra-core
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faster-whisper
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tensorboard
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natsort
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einops
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lightning
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hydra-core
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faster-whisper
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+
qwen-asr
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tensorboard
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natsort
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einops
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