Update custom_interface_app.py
Browse files- custom_interface_app.py +36 -13
custom_interface_app.py
CHANGED
@@ -133,10 +133,11 @@ class ASR(Pretrained):
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# Get audio length in seconds
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audio_length = len(waveform) / 16000
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if audio_length >=
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# split audio every 20 seconds
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segments = []
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num_segments = int(np.ceil(len(waveform) / max_duration))
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start = 0
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for i in range(num_segments):
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@@ -159,7 +160,14 @@ class ASR(Pretrained):
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# Pass the segment through the ASR model
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segment_output = self.encode_batch_w2v2(device, batch, rel_length)
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else:
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waveform = torch.tensor(waveform).to(device)
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waveform = waveform.to(device)
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@@ -167,7 +175,7 @@ class ASR(Pretrained):
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batch = waveform.unsqueeze(0)
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rel_length = torch.tensor([1.0]).to(device)
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outputs = self.encode_batch_w2v2(device, batch, rel_length)
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@@ -179,10 +187,11 @@ class ASR(Pretrained):
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# Get audio length in seconds
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audio_length = len(waveform) / 16000
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if audio_length >=
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# split audio every 20 seconds
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segments = []
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num_segments = int(np.ceil(len(waveform) / max_duration))
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start = 0
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for i in range(num_segments):
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@@ -205,21 +214,28 @@ class ASR(Pretrained):
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# Pass the segment through the ASR model
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segment_output = self.encode_batch_whisper(device, batch, rel_length)
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else:
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waveform = torch.tensor(waveform).to(device)
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waveform = waveform.to(device)
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batch = waveform.unsqueeze(0)
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rel_length = torch.tensor([1.0]).to(device)
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outputs = self.encode_batch_whisper(device, batch, rel_length)
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def classify_file_whisper(self, waveform, pipe, device):
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# waveform, sr = librosa.load(path, sr=16000)
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transcription = pipe(waveform, generate_kwargs={"language": "macedonian"})["text"]
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return transcription
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def classify_file_mms(self, waveform, processor, model, device):
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@@ -229,10 +245,11 @@ class ASR(Pretrained):
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# Get audio length in seconds
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audio_length = len(waveform) / 16000
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if audio_length >=
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# split audio every 20 seconds
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segments = []
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-
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num_segments = int(np.ceil(len(waveform) / max_duration))
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start = 0
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for i in range(num_segments):
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@@ -255,7 +272,13 @@ class ASR(Pretrained):
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outputs = model(**inputs).logits
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ids = torch.argmax(outputs, dim=-1)[0]
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segment_output = processor.decode(ids)
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else:
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waveform = torch.tensor(waveform).to(device)
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inputs = processor(waveform, sampling_rate=16_000, return_tensors="pt").to(device)
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@@ -263,4 +286,4 @@ class ASR(Pretrained):
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outputs = model(**inputs).logits
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ids = torch.argmax(outputs, dim=-1)[0]
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transcription = processor.decode(ids)
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# Get audio length in seconds
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audio_length = len(waveform) / 16000
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if audio_length >= 30:
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# split audio every 20 seconds
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segments = []
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all_segments = []
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max_duration = 30 * 16000 # Maximum segment duration in samples (20 seconds)
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num_segments = int(np.ceil(len(waveform) / max_duration))
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start = 0
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for i in range(num_segments):
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# Pass the segment through the ASR model
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segment_output = self.encode_batch_w2v2(device, batch, rel_length)
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segment_output = [" ".join(segment) for segment in segment_output]
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all_segments.append(segment_output)
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segments = ""
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for segment in all_segments:
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segment = segment[0]
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segments += segment + " "
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return [segments]
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else:
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waveform = torch.tensor(waveform).to(device)
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waveform = waveform.to(device)
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batch = waveform.unsqueeze(0)
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rel_length = torch.tensor([1.0]).to(device)
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outputs = self.encode_batch_w2v2(device, batch, rel_length)
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return [" ".join(out) for out in outputs]
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# Get audio length in seconds
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audio_length = len(waveform) / 16000
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if audio_length >= 30:
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# split audio every 20 seconds
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segments = []
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all_segments = []
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max_duration = 30 * 16000 # Maximum segment duration in samples (20 seconds)
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num_segments = int(np.ceil(len(waveform) / max_duration))
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start = 0
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for i in range(num_segments):
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# Pass the segment through the ASR model
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segment_output = self.encode_batch_whisper(device, batch, rel_length)
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# segment_output = [" ".join(segment) for segment in segment_output]
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all_segments.append(segment_output)
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segments = ""
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for segment in all_segments:
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segment = segment[0]
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segments += segment + " "
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return [segments]
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else:
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waveform = torch.tensor(waveform).to(device)
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waveform = waveform.to(device)
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batch = waveform.unsqueeze(0)
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rel_length = torch.tensor([1.0]).to(device)
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outputs = self.encode_batch_whisper(device, batch, rel_length)
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return outputs
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def classify_file_whisper(self, waveform, pipe, device):
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# waveform, sr = librosa.load(path, sr=16000)
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transcription = pipe(waveform, generate_kwargs={"language": "macedonian"})["text"]
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return [transcription]
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def classify_file_mms(self, waveform, processor, model, device):
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# Get audio length in seconds
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audio_length = len(waveform) / 16000
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if audio_length >= 30:
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# split audio every 20 seconds
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segments = []
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all_segments = []
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max_duration = 30 * 16000 # Maximum segment duration in samples (20 seconds)
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num_segments = int(np.ceil(len(waveform) / max_duration))
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start = 0
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for i in range(num_segments):
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outputs = model(**inputs).logits
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ids = torch.argmax(outputs, dim=-1)[0]
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segment_output = processor.decode(ids)
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# segment_output = [" ".join(segment) for segment in segment_output]
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all_segments.append(segment_output)
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segments = ""
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for segment in all_segments:
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segments += segment + " "
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return [segments]
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else:
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waveform = torch.tensor(waveform).to(device)
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inputs = processor(waveform, sampling_rate=16_000, return_tensors="pt").to(device)
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outputs = model(**inputs).logits
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ids = torch.argmax(outputs, dim=-1)[0]
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transcription = processor.decode(ids)
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return [transcription]
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