Spaces:
Sleeping
Sleeping
Aryan Wadhawan
commited on
Commit
·
e59b0bd
1
Parent(s):
a7fd32e
Implemented everything
Browse files- app.py +61 -7
- requirements.txt +2 -1
app.py
CHANGED
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@@ -3,23 +3,31 @@ from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC
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import torch
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import phonemizer
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import librosa
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import io
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import base64
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def lark(audioAsB64):
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# base64 to wav data conversion
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wav_data = base64.b64decode(audioAsB64.encode("utf-8"))
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#
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processor = Wav2Vec2Processor.from_pretrained(
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"facebook/wav2vec2-xlsr-53-espeak-cv-ft"
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)
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model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-xlsr-53-espeak-cv-ft")
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waveform, sample_rate = librosa.load(
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io.BytesIO(wav_data), sr=16000
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) # Downsample 44.1kHz to 8kHz
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input_values = processor(
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waveform, sampling_rate=sample_rate, return_tensors="pt"
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@@ -29,10 +37,56 @@ def lark(audioAsB64):
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logits = model(input_values).logits
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predicted_ids = torch.argmax(logits, dim=-1)
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return
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iface = gr.Interface(fn=lark, inputs="text", outputs="text")
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iface.launch()
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import torch
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import phonemizer
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import librosa
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import math
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import io
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import base64
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from strsimpy.jaro_winkler import JaroWinkler
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# base64 to audio ✅
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# audio to transcription ✅
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# audio to text ✅
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# text to phoneme ✅
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# accuracy = jarowinkler(transcription, phoneme) ✅
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# band = getBandFromAccuracy(accuracy) ✅
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# return accuracy, band ✅
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def lark(audioAsB64):
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# base64 to wav data conversion
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wav_data = base64.b64decode(audioAsB64.encode("utf-8"))
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# audio to transcription
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processor = Wav2Vec2Processor.from_pretrained(
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"facebook/wav2vec2-xlsr-53-espeak-cv-ft"
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)
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model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-xlsr-53-espeak-cv-ft")
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waveform, sample_rate = librosa.load(io.BytesIO(wav_data), sr=16000)
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input_values = processor(
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waveform, sampling_rate=sample_rate, return_tensors="pt"
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logits = model(input_values).logits
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predicted_ids = torch.argmax(logits, dim=-1)
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speechToPhonemeTranscription = processor.batch_decode(predicted_ids)[0]
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# audio to text
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processorSTT = Wav2Vec2Processor.from_pretrained("facebook/wav2vec2-base-960h")
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model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-base-960h")
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input_values = processorSTT(
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waveform, sampling_rate=sample_rate, return_tensors="pt"
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).input_values
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logits = model(input_values).logits
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predicted_ids = torch.argmax(logits, dim=-1)
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speechToTextTranscripition = processor.batch_decode(predicted_ids)[0]
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# text to phoneme
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graphemeToPhonemeTranscription = phonemizer.phonemize(speechToTextTranscripition)
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# accuracy = jaroWinkler(transcription, phoneme)
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jarowinkler = JaroWinkler()
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similarity_score = jarowinkler.similarity(
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speechToPhonemeTranscription, graphemeToPhonemeTranscription
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)
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# ielts pronunciation band estimation
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def getBandFromSimilarityScore(similarity_score):
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if similarity_score >= 0.91:
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return 9
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elif similarity_score >= 0.81:
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return 8
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elif similarity_score >= 0.73:
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return 7
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elif similarity_score >= 0.65:
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return 6
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elif similarity_score >= 0.60:
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return 5
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elif similarity_score >= 0.46:
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return 4
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elif similarity_score >= 0.35:
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return 3
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elif similarity_score >= 0.1:
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return 2
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else:
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return 1
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IELTSband = getBandFromSimilarityScore(similarity_score)
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return [similarity_score, IELTSband]
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iface = gr.Interface(fn=lark, inputs="text", outputs=["text", "text"])
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iface.launch()
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requirements.txt
CHANGED
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@@ -1,4 +1,5 @@
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phonemizer
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librosa
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transformers
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-
torch
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phonemizer
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librosa
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transformers
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+
torch
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strsimpy
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