Create custom_interface_app.py

custom_interface_app.py

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+import torch
+from speechbrain.inference.interfaces import Pretrained
+import librosa
+import numpy as np
+
+
+class ASR(Pretrained):
+    def __init__(self, *args, **kwargs):
+        super().__init__(*args, **kwargs)
+
+    def encode_batch(self, device, wavs, wav_lens=None, normalize=False):
+        wavs = wavs.to(device)
+        wav_lens = wav_lens.to(device)
+
+        # Forward pass
+        encoded_outputs = self.mods.encoder_w2v2(wavs.detach())
+        # append
+        tokens_bos = torch.zeros((wavs.size(0), 1), dtype=torch.long).to(device)
+        embedded_tokens = self.mods.embedding(tokens_bos)
+        decoder_outputs, _ = self.mods.decoder(embedded_tokens, encoded_outputs, wav_lens)
+
+        # Output layer for seq2seq log-probabilities
+        predictions = self.hparams.test_search(encoded_outputs, wav_lens)[0]
+        # predicted_words = [self.hparams.tokenizer.decode_ids(prediction).split(" ") for prediction in predictions]
+        predicted_words = []
+        for prediction in predictions:
+            prediction = [token for token in prediction if token != 0]
+            predicted_words.append(self.hparams.tokenizer.decode_ids(prediction).split(" "))
+        prediction = []
+        for sent in predicted_words:
+            sent = self.filter_repetitions(sent, 3)
+            prediction.append(sent)
+        predicted_words = prediction
+        return predicted_words
+
+    def filter_repetitions(self, seq, max_repetition_length):
+        seq = list(seq)
+        output = []
+        max_n = len(seq) // 2
+        for n in range(max_n, 0, -1):
+            max_repetitions = max(max_repetition_length // n, 1)
+            # Don't need to iterate over impossible n values:
+            # len(seq) can change a lot during iteration
+            if (len(seq) <= n*2) or (len(seq) <= max_repetition_length):
+                continue
+            iterator = enumerate(seq)
+            # Fill first buffers:
+            buffers = [[next(iterator)[1]] for _ in range(n)]
+            for seq_index, token in iterator:
+                current_buffer = seq_index % n
+                if token != buffers[current_buffer][-1]:
+                    # No repeat, we can flush some tokens
+                    buf_len = sum(map(len, buffers))
+                    flush_start = (current_buffer-buf_len) % n
+                    # Keep n-1 tokens, but possibly mark some for removal
+                    for flush_index in range(buf_len - buf_len%n):
+                        if (buf_len - flush_index) > n-1:
+                            to_flush = buffers[(flush_index + flush_start) % n].pop(0)
+                        else:
+                            to_flush = None
+                        # Here, repetitions get removed:
+                        if (flush_index // n < max_repetitions) and to_flush is not None:
+                            output.append(to_flush)
+                        elif (flush_index // n >= max_repetitions) and to_flush is None:
+                            output.append(to_flush)
+                buffers[current_buffer].append(token)
+            # At the end, final flush
+            current_buffer += 1
+            buf_len = sum(map(len, buffers))
+            flush_start = (current_buffer-buf_len) % n
+            for flush_index in range(buf_len):
+                to_flush = buffers[(flush_index + flush_start) % n].pop(0)
+                # Here, repetitions just get removed:
+                if flush_index // n < max_repetitions:
+                    output.append(to_flush)
+            seq = []
+            to_delete = 0
+            for token in output:
+                if token is None:
+                    to_delete += 1
+                elif to_delete > 0:
+                    to_delete -= 1
+                else:
+                    seq.append(token)
+            output = []
+        return seq
+    
+
+    def increase_volume(self, waveform, threshold_db=-25):
+        # Measure loudness using RMS
+        loudness_vector = librosa.feature.rms(y=waveform)
+        average_loudness = np.mean(loudness_vector)
+        average_loudness_db = librosa.amplitude_to_db(average_loudness)
+
+        print(f"Average Loudness: {average_loudness_db} dB")
+
+        # Check if loudness is below threshold and apply gain if needed
+        if average_loudness_db < threshold_db:
+            # Calculate gain needed
+            gain_db = threshold_db - average_loudness_db
+            gain = librosa.db_to_amplitude(gain_db)  # Convert dB to amplitude factor
+
+            # Apply gain to the audio signal
+            waveform = waveform * gain
+            loudness_vector = librosa.feature.rms(y=waveform)
+            average_loudness = np.mean(loudness_vector)
+            average_loudness_db = librosa.amplitude_to_db(average_loudness)
+
+            print(f"Average Loudness: {average_loudness_db} dB")
+        return waveform
+
+
+    def classify_file(self, path, device):
+        # Load the audio file
+        # path = "long_sample.wav"
+        waveform, sr = librosa.load(path, sr=16000)
+
+        # increase the volume if needed
+        # waveform = self.increase_volume(waveform)
+
+        # Get audio length in seconds
+        audio_length = len(waveform) / sr
+        
+        if audio_length >= 20:
+            print(f"Audio is too long ({audio_length:.2f} seconds), splitting into segments")
+            # Detect non-silent segments
+
+            non_silent_intervals = librosa.effects.split(waveform, top_db=20)  # Adjust top_db for sensitivity
+
+            segments = []
+            current_segment = []
+            current_length = 0
+            max_duration = 20 * sr  # Maximum segment duration in samples (20 seconds)
+
+
+            for interval in non_silent_intervals:
+                start, end = interval
+                segment_part = waveform[start:end]
+
+                # If adding the next part exceeds max duration, store the segment and start a new one
+                if current_length + len(segment_part) > max_duration:
+                    segments.append(np.concatenate(current_segment))
+                    current_segment = []
+                    current_length = 0
+
+                current_segment.append(segment_part)
+                current_length += len(segment_part)
+
+            # Append the last segment if it's not empty
+            if current_segment:
+                segments.append(np.concatenate(current_segment))
+
+            # Process each segment
+            outputs = []
+            for i, segment in enumerate(segments):
+                print(f"Processing segment {i + 1}/{len(segments)}, length: {len(segment) / sr:.2f} seconds")
+
+                # import soundfile as sf
+                # sf.write(f"outputs/segment_{i}.wav", segment, sr)
+
+                segment_tensor = torch.tensor(segment).to(device)
+
+                # Fake a batch for the segment
+                batch = segment_tensor.unsqueeze(0).to(device)
+                rel_length = torch.tensor([1.0]).to(device)  # Adjust if necessary
+
+                # Pass the segment through the ASR model
+                segment_output = self.encode_batch(device, batch, rel_length)
+                yield segment_output
+        else:
+            waveform = torch.tensor(waveform).to(device)
+            waveform = waveform.to(device)
+            # Fake a batch:
+            batch = waveform.unsqueeze(0)
+            rel_length = torch.tensor([1.0]).to(device)
+            outputs = self.encode_batch(device, batch, rel_length)
+            yield outputs
+
+
+    def classify_file_whisper(self, path, pipe, device):
+        waveform, sr = librosa.load(path, sr=16000)
+        transcription = pipe(waveform, generate_kwargs={"language": "macedonian"})["text"]
+        return transcription
+       
+
+    def classify_file_mms(self, path, processor, model, device):
+        # Load the audio file
+        waveform, sr = librosa.load(path, sr=16000)
+
+        # Get audio length in seconds
+        audio_length = len(waveform) / sr
+        
+        if audio_length >= 20:
+            print(f"MMS Audio is too long ({audio_length:.2f} seconds), splitting into segments")
+            # Detect non-silent segments
+            non_silent_intervals = librosa.effects.split(waveform, top_db=20)  # Adjust top_db for sensitivity
+
+            segments = []
+            current_segment = []
+            current_length = 0
+            max_duration = 20 * sr  # Maximum segment duration in samples (20 seconds)
+
+
+            for interval in non_silent_intervals:
+                start, end = interval
+                segment_part = waveform[start:end]
+
+                # If adding the next part exceeds max duration, store the segment and start a new one
+                if current_length + len(segment_part) > max_duration:
+                    segments.append(np.concatenate(current_segment))
+                    current_segment = []
+                    current_length = 0
+
+                current_segment.append(segment_part)
+                current_length += len(segment_part)
+
+            # Append the last segment if it's not empty
+            if current_segment:
+                segments.append(np.concatenate(current_segment))
+
+            # Process each segment
+            outputs = []
+            for i, segment in enumerate(segments):
+                print(f"MMS Processing segment {i + 1}/{len(segments)}, length: {len(segment) / sr:.2f} seconds")
+
+                segment_tensor = torch.tensor(segment).to(device)
+
+                # Pass the segment through the ASR model
+                inputs = processor(segment_tensor, sampling_rate=16_000, return_tensors="pt").to(device)
+                outputs = model(**inputs).logits
+                ids = torch.argmax(outputs, dim=-1)[0]
+                segment_output = processor.decode(ids)
+                yield segment_output
+        else:
+            waveform = torch.tensor(waveform).to(device)
+            inputs = processor(waveform, sampling_rate=16_000, return_tensors="pt").to(device)
+            outputs = model(**inputs).logits
+            ids = torch.argmax(outputs, dim=-1)[0]
+            transcription = processor.decode(ids)
+            yield transcription