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Running
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Irpan
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Commit
·
3a18b3b
1
Parent(s):
1959ce1
asr
Browse files
app.py
CHANGED
@@ -1,11 +1,17 @@
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import gradio as gr
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# from tts import synthesize
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mms_transcribe = gr.Interface(
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fn=transcribe,
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inputs=[
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gr.Audio()
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],
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outputs="text",
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import gradio as gr
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import asr
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# from tts import synthesize
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mms_transcribe = gr.Interface(
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fn=asr.transcribe,
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inputs=[
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gr.Dropdown(
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choices=[m["id"] for m in asr.models_info],
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label="Select Model for ASR",
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value="ixxan/wav2vec2-large-mms-1b-uyghur-latin",
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interactive=True
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),
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gr.Audio()
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],
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outputs="text",
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asr.py
CHANGED
@@ -1,15 +1,45 @@
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import torchaudio
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import torch
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from transformers import
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import numpy as np
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# Load processor and model
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def transcribe(audio_data) -> str:
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"""
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Transcribes audio to text using the Whisper model for Uyghur.
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Args:
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@@ -35,13 +65,18 @@ def transcribe(audio_data) -> str:
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return "<<ERROR: Invalid Audio Input Instance: {}>>".format(type(audio_data))
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# Resample if needed
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if sampling_rate != target_sr:
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resampler = torchaudio.transforms.Resample(sampling_rate, target_sr)
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audio_input = resampler(audio_input)
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# Preprocess the audio input
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inputs = processor(audio_input.squeeze(), sampling_rate=target_sr, return_tensors="pt")
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# Move model to GPU if available
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device = "cuda" if torch.cuda.is_available() else "cpu"
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# Generate transcription
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with torch.no_grad():
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return transcription
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import torchaudio
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import torch
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from transformers import (
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WhisperProcessor,
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AutoProcessor,
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AutoModelForSpeechSeq2Seq,
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AutoModelForCTC,
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Wav2Vec2Processor,
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Wav2Vec2ForCTC
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)
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import numpy as np
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# Load processor and model
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models_info = {
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"openai/whisper-small-uzbek": {
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"processor": WhisperProcessor.from_pretrained("openai/whisper-small", language="uzbek", task="transcribe"),
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"model": AutoModelForSpeechSeq2Seq.from_pretrained("openai/whisper-small"),
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"ctc_model": False
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},
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"ixxan/whisper-small-thugy20": {
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"processor": AutoProcessor.from_pretrained("ixxan/whisper-small-thugy20"),
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"model": AutoModelForSpeechSeq2Seq.from_pretrained("ixxan/whisper-small-thugy20"),
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"ctc_model": False
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},
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"ixxan/whisper-small-uyghur-common-voice": {
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"processor": AutoProcessor.from_pretrained("ixxan/whisper-small-uyghur-common-voice"),
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"model": AutoModelForSpeechSeq2Seq.from_pretrained("ixxan/whisper-small-uyghur-common-voice"),
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"ctc_model": False
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},
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"facebook/mms-1b-all": {
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"processor": AutoProcessor.from_pretrained("facebook/mms-1b-all", target_lang='uig-script_arabic'),
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"model": AutoModelForCTC.from_pretrained("facebook/mms-1b-all", target_lang='uig-script_arabic', ignore_mismatched_sizes=True),
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"ctc_model": True
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},
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# "ixxan/wav2vec2-large-mms-1b-uyghur-latin": {
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# "processor": Wav2Vec2Processor.from_pretrained("ixxan/wav2vec2-large-mms-1b-uyghur-latin", target_lang='uig-script_latin'),
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# "model": Wav2Vec2ForCTC.from_pretrained("ixxan/wav2vec2-large-mms-1b-uyghur-latin"),
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# "ctc_model": True
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# },
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}
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def transcribe(audio_data, model_id) -> str:
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"""
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Transcribes audio to text using the Whisper model for Uyghur.
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Args:
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return "<<ERROR: Invalid Audio Input Instance: {}>>".format(type(audio_data))
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model = models_info[model_id]["model"]
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processor = models_info[model_id]["processor"]
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target_sr = processor.feature_extractor.sampling_rate
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ctc_model = models_info[model_id]["ctc_model"]
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# Resample if needed
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if sampling_rate != target_sr:
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resampler = torchaudio.transforms.Resample(sampling_rate, target_sr)
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audio_input = resampler(audio_input)
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# Preprocess the audio input
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inputs = processor(audio_input.squeeze(), sampling_rate=target_sr, return_tensors="pt", padding=True)
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# Move model to GPU if available
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device = "cuda" if torch.cuda.is_available() else "cpu"
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# Generate transcription
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with torch.no_grad():
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if ctc_model:
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logits = model(**inputs).logits
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predicted_ids = torch.argmax(logits, dim=-1)
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transcription = processor.batch_decode(predicted_ids)[0]
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else:
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generated_ids = model.generate(inputs["input_features"], max_length=225)
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transcription = processor.batch_decode(generated_ids, skip_special_tokens=True)[0]
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return transcription
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