Create audio_processing.py, include module for language detection, modifying sampling rate and processing audios longer than 30 seconds

audio_processing.py

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+import torch
+import whisper
+import torchaudio as ta
+from model_utils import get_processor, get_model, get_whisper_model_small, get_device
+from config import SAMPLING_RATE, CHUNK_LENGTH_S
+
+def detect_language(audio_file):
+    whisper_model = get_whisper_model_small()
+    trimmed_audio = whisper.pad_or_trim(audio_file.squeeze())
+    mel = whisper.log_mel_spectrogram(trimmed_audio).to(whisper_model.device)
+    _, probs = whisper_model.detect_language(mel)
+    detected_lang = max(probs[0], key=probs[0].get)
+    print(f"Detected language: {detected_lang}")
+    return detected_lang
+
+def process_long_audio(waveform, sampling_rate, task="transcribe", language=None):
+    processor = get_processor()
+    model = get_model()
+    device = get_device()
+
+    input_length = waveform.shape[1]
+    chunk_length = int(CHUNK_LENGTH_S * sampling_rate)
+    chunks = [waveform[:, i:i + chunk_length] for i in range(0, input_length, chunk_length)]
+
+    results = []
+    for chunk in chunks:
+        input_features = processor(chunk[0], sampling_rate=sampling_rate, return_tensors="pt").input_features.to(device)
+
+        with torch.no_grad():
+            if task == "translate":
+                forced_decoder_ids = processor.get_decoder_prompt_ids(language=language, task="translate")
+                generated_ids = model.generate(input_features, forced_decoder_ids=forced_decoder_ids)
+            else:
+                generated_ids = model.generate(input_features)
+
+        transcription = processor.batch_decode(generated_ids, skip_special_tokens=True)
+        results.extend(transcription)
+
+        # Clear GPU cache
+        torch.cuda.empty_cache()
+
+    return " ".join(results)
+
+def load_and_resample_audio(file):
+    waveform, sampling_rate = ta.load(file)
+    if sampling_rate != SAMPLING_RATE:
+        waveform = ta.functional.resample(waveform, orig_freq=sampling_rate, new_freq=SAMPLING_RATE)
+    return waveform