Update app.py
Browse files
app.py
CHANGED
@@ -6,10 +6,14 @@ import requests
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from gtts import gTTS
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from transformers import AutoProcessor, AutoModelForSpeechSeq2Seq
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from pydub import AudioSegment
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RAPIDAPI_KEY =
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GROQ_API_KEY =
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# Load the Whisper model
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processor = AutoProcessor.from_pretrained("ihanif/whisper-medium-urdu")
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@@ -43,10 +47,21 @@ def process_audio(file_path):
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# Convert audio to numpy array for processing
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audio_samples = np.array(audio.get_array_of_samples(), dtype=np.float32) / 32768.0 # Normalize to [-1, 1] range
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audio_input = processor(audio_samples, return_tensors="pt", sampling_rate=16000)
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# Transcribe the audio using the fine-tuned Whisper model
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text = processor.batch_decode(result, skip_special_tokens=True)[0]
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if not text.strip(): # Check if the transcribed text is empty
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@@ -58,22 +73,15 @@ def process_audio(file_path):
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urdu_to_eng = translate("en", text)
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print(f"Translated Text (English): {urdu_to_eng}") # Debugging step
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#
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"
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"max_tokens": 50
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}
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response = requests.post(groq_url, json=groq_payload, headers=groq_headers)
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chat_completion = response.json()
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# Access the response
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response_message = chat_completion["choices"][0]["message"]["content"].strip()
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print(f"Groq Response (English): {response_message}") # Debugging step
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# Translate the response text back to Urdu
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@@ -104,4 +112,4 @@ iface = gr.Interface(
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live=True
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iface.launch()
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from gtts import gTTS
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from transformers import AutoProcessor, AutoModelForSpeechSeq2Seq
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from pydub import AudioSegment
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from groq import Groq
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from google.colab import userdata
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RAPIDAPI_KEY = userdata.get('RAPIDAPI_LANG_TRANS')
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GROQ_API_KEY = userdata.get('GROQ_API_KEY')
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# Initialize the Groq client
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client = Groq(api_key=GROQ_API_KEY)
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# Load the Whisper model
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processor = AutoProcessor.from_pretrained("ihanif/whisper-medium-urdu")
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# Convert audio to numpy array for processing
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audio_samples = np.array(audio.get_array_of_samples(), dtype=np.float32) / 32768.0 # Normalize to [-1, 1] range
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# Create attention mask
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# Assume padding length is determined by the maximum length of sequences
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# For simplicity, we'll just create a mask where all values are 1 (no padding)
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# In practice, you would adjust this based on actual sequence length
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attention_mask = np.ones_like(audio_samples, dtype=np.int64)
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audio_input = processor(audio_samples, return_tensors="pt", sampling_rate=16000)
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# Transcribe the audio using the fine-tuned Whisper model
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# Pass the attention mask as well
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result = model.generate(
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**audio_input,
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attention_mask=torch.tensor(attention_mask).unsqueeze(0) # Add batch dimension
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)
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text = processor.batch_decode(result, skip_special_tokens=True)[0]
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if not text.strip(): # Check if the transcribed text is empty
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urdu_to_eng = translate("en", text)
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print(f"Translated Text (English): {urdu_to_eng}") # Debugging step
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# Generate a response using Groq
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chat_completion = client.chat.completions.create(
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messages=[{"role": "user", "content": urdu_to_eng}],
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model="llama3-8b-8192", # Ensure the model supports Urdu if possible
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max_tokens=50
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)
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# Access the response using dot notation
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response_message = chat_completion.choices[0].message.content.strip()
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print(f"Groq Response (English): {response_message}") # Debugging step
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# Translate the response text back to Urdu
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live=True
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)
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iface.launch(share=True)
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