Update app.py
Browse files
app.py
CHANGED
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@@ -1,13 +1,13 @@
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# ==============================================================================
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#
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# ==============================================================================
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# This script is
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#
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#
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# KEY FIX:
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#
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#
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#
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# ==============================================================================
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import gradio as gr
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@@ -19,20 +19,19 @@ import numpy as np
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import soundfile as sf
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from pydub import AudioSegment, effects
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# ---
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from transformers import SpeechT5Processor, SpeechT5ForTextToSpeech, SpeechT5HifiGan
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from speechbrain.pretrained import EncoderClassifier
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# --- Model Loading ---
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print("Loading models, this may take a moment...")
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device = "cuda" if torch.cuda.is_available() else "cpu"
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print(f"Using device: {device}")
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processor = SpeechT5Processor.from_pretrained("microsoft/speecht5_tts")
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model = SpeechT5ForTextToSpeech.from_pretrained("Somalitts/8aad").to(device)
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vocoder = SpeechT5HifiGan.from_pretrained("microsoft/speecht5_hifigan").to(device)
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speaker_model = EncoderClassifier.from_hparams(
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source="speechbrain/spkrec-xvect-voxceleb",
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run_opts={"device": device},
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@@ -41,38 +40,26 @@ speaker_model = EncoderClassifier.from_hparams(
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print("Models loaded successfully.")
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# --- Speaker Embedding
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# The quality of
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#
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# For best results, your `1.wav` file should be:
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# - At least 30 seconds long.
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# - Contain clear speech with NO background noise or echo.
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# - Contain only one speaker.
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def create_speaker_embedding(audio_path):
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print("Creating speaker embedding from:", audio_path)
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waveform, sr = torchaudio.load(audio_path)
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# Resample to 16000 Hz if it's not already
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if sr != 16000:
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with torch.no_grad():
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embedding = speaker_model.encode_batch(waveform.to(device))
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# Normalize the embedding for the TTS model
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embedding = torch.nn.functional.normalize(embedding, dim=2).squeeze()
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print("Speaker embedding created.")
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return embedding
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SPEAKER_WAV = "1.wav"
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EMB_PATH = "speaker_embedding.pt"
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# For Hugging Face Spaces, create a dummy file if `1.wav` doesn't exist
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if not os.path.exists(SPEAKER_WAV):
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print(f"Warning: '{SPEAKER_WAV}' not found. Creating a silent dummy file.
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sf.write(SPEAKER_WAV, np.zeros(16000 * 2), 16000)
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# Generate and cache the speaker embedding
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print("Loading or creating speaker embedding...")
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speaker_embedding = create_speaker_embedding(SPEAKER_WAV)
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print("Speaker embedding ready.")
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@@ -82,14 +69,11 @@ print("Speaker embedding ready.")
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def number_to_somali_words(num_str):
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try:
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num = int(num_str)
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except ValueError:
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return num_str # Not a number
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if num < 0: return "eber ka yar"
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units = ["", "koow", "labo", "saddex", "afar", "shan", "lix", "toddobo", "siddeed", "sagaal"]
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teens = ["toban", "kow iyo toban", "laba iyo toban", "saddex iyo toban", "afar iyo toban", "shan iyo toban", "lix iyo toban", "toddobo iyo toban", "siddeed iyo toban", "sagaal iyo toban"]
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tens = ["", "toban", "labaatan", "soddon", "afartan", "konton", "lixdan", "toddobaatan", "siddeetan", "sagaashan"]
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if num == 0: return "eber"
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if num < 10: return units[num]
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if num < 20: return teens[num-10]
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@@ -111,42 +95,51 @@ def normalize_text(text):
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return text.strip()
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# --- Core TTS Function with
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def text_to_speech(text):
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print(f"Generating speech for: '{text}'")
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normalized_text = normalize_text(text)
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if not normalized_text:
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return (16000, np.zeros(0, dtype=np.int16))
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inputs = processor(text=normalized_text, return_tensors="pt").to(device)
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with torch.no_grad():
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# --- QUALITY IMPROVEMENT ---
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# The `threshold` parameter helps the model stop generating more naturally.
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# This is a key factor in reducing robotic artifacts.
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speech = model.generate_speech(
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inputs["input_ids"],
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speaker_embedding.unsqueeze(0),
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vocoder=vocoder,
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threshold=0.5
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)
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# Post-processing: Normalize volume for a polished feel.
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# This does not fix distortion, but it improves listenability.
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audio_segment = AudioSegment(
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frame_rate=16000,
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sample_width=
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channels=1
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)
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processed_audio = effects.normalize(audio_segment)
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print("Speech generation
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return (16000,
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# --- Gradio Web Interface ---
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fn=text_to_speech,
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inputs=gr.Textbox(
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label="Qoraalka Geli (Enter Somali Text)",
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placeholder="Ku qor qoraalkaaga halkan..."
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),
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outputs=gr.Audio(
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label="Codka La Soo Saaray (
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type="numpy"
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),
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title="🇸🇴 Somali TTS
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description=(
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"Ku qor qoraal si aad ugu
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"\n\n(Enter text to convert to a
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),
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examples=[
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["Sidee tahay saaxiib? Maanta waa maalin qurux badan."],
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["
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["
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]
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)
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# ==============================================================================
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# Somali TTS with AI-Powered Noise Reduction
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# ==============================================================================
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# This script is the final version, designed to produce a clean, studio-quality
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# voice by removing background noise and digital artifacts.
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#
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# KEY FIX:
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# A noise reduction filter (`noisereduce`) is applied directly to the
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# generated audio. This intelligently removes hiss and unwanted noise,
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# leaving only the clean voice.
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# ==============================================================================
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import gradio as gr
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import soundfile as sf
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from pydub import AudioSegment, effects
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# --- Required Imports for TTS and Noise Reduction ---
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from transformers import SpeechT5Processor, SpeechT5ForTextToSpeech, SpeechT5HifiGan
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from speechbrain.pretrained import EncoderClassifier
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import noisereduce as nr # Import the noise reduction library
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# --- Model Loading ---
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print("Loading models, this may take a moment...")
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device = "cuda" if torch.cuda.is_available() else "cpu"
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print(f"Using device: {device}")
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processor = SpeechT5Processor.from_pretrained("microsoft/speecht5_tts")
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model = SpeechT5ForTextToSpeech.from_pretrained("Somalitts/8aad").to(device)
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vocoder = SpeechT5HifiGan.from_pretrained("microsoft/speecht5_hifigan").to(device)
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speaker_model = EncoderClassifier.from_hparams(
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source="speechbrain/spkrec-xvect-voxceleb",
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run_opts={"device": device},
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print("Models loaded successfully.")
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# --- Speaker Embedding Generation ---
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# The quality of your `1.wav` file is CRITICAL for good results.
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# It should be a clean, noise-free recording of a single speaker.
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def create_speaker_embedding(audio_path):
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print("Creating speaker embedding from:", audio_path)
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waveform, sr = torchaudio.load(audio_path)
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if sr != 16000:
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waveform = torchaudio.functional.resample(waveform, sr, 16000)
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with torch.no_grad():
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embedding = speaker_model.encode_batch(waveform.to(device))
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embedding = torch.nn.functional.normalize(embedding, dim=2).squeeze()
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print("Speaker embedding created.")
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return embedding
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SPEAKER_WAV = "1.wav"
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if not os.path.exists(SPEAKER_WAV):
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print(f"Warning: '{SPEAKER_WAV}' not found. Creating a silent dummy file.")
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sf.write(SPEAKER_WAV, np.zeros(16000 * 2), 16000)
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print("Loading or creating speaker embedding...")
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speaker_embedding = create_speaker_embedding(SPEAKER_WAV)
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print("Speaker embedding ready.")
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def number_to_somali_words(num_str):
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try:
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num = int(num_str)
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except ValueError: return num_str
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if num < 0: return "eber ka yar"
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units = ["", "koow", "labo", "saddex", "afar", "shan", "lix", "toddobo", "siddeed", "sagaal"]
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teens = ["toban", "kow iyo toban", "laba iyo toban", "saddex iyo toban", "afar iyo toban", "shan iyo toban", "lix iyo toban", "toddobo iyo toban", "siddeed iyo toban", "sagaal iyo toban"]
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tens = ["", "toban", "labaatan", "soddon", "afartan", "konton", "lixdan", "toddobaatan", "siddeetan", "sagaashan"]
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if num == 0: return "eber"
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if num < 10: return units[num]
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if num < 20: return teens[num-10]
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return text.strip()
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# --- Core TTS Function with AI Noise Reduction ---
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def text_to_speech(text):
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print(f"Generating speech for: '{text}'")
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normalized_text = normalize_text(text)
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if not normalized_text:
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return (16000, np.zeros(0, dtype=np.int16))
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# --- Step 1: Generate the raw speech ---
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inputs = processor(text=normalized_text, return_tensors="pt").to(device)
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with torch.no_grad():
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speech = model.generate_speech(
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inputs["input_ids"],
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speaker_embedding.unsqueeze(0),
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vocoder=vocoder,
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threshold=0.5
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)
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raw_speech_numpy = speech.cpu().numpy()
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# --- Step 2: AI-POWERED NOISE REDUCTION ---
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# This is the crucial step to clean the audio.
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print("Applying noise reduction filter...")
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# The sample rate (sr) must match the audio's sample rate.
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clean_speech = nr.reduce_noise(y=raw_speech_numpy, sr=16000)
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print("Noise reduction complete.")
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# --- Step 3: Final Polishing (Volume Normalization) ---
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# Convert to pydub AudioSegment for easy volume handling.
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# Note: Ensure the numpy array is in 16-bit integer format for pydub.
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clean_speech_int16 = (clean_speech * 32767).astype(np.int16)
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audio_segment = AudioSegment(
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clean_speech_int16.tobytes(),
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frame_rate=16000,
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sample_width=clean_speech_int16.dtype.itemsize,
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channels=1
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)
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# Normalize volume to a standard level for a professional feel.
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processed_audio = effects.normalize(audio_segment)
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# Convert back to numpy array for Gradio output
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final_output_numpy = np.array(processed_audio.get_array_of_samples())
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print("Speech generation and cleaning finished.")
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return (16000, final_output_numpy)
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# --- Gradio Web Interface ---
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fn=text_to_speech,
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inputs=gr.Textbox(
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label="Qoraalka Geli (Enter Somali Text)",
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placeholder="Ku qor qoraalkaaga halkan si aad u hesho cod saafi ah..."
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),
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outputs=gr.Audio(
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label="Codka La Soo Saaray (Cleaned Audio)",
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type="numpy"
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),
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title="🇸🇴 Somali TTS oo leh Cod Sifeeye (with Noise Reduction)",
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description=(
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"Ku qor qoraal si aad ugu beddesho cod saafi ah oo aan qaylo lahayn. Barnaamijkan wuxuu si toos ah uga saarayaa sawaxanka codka la soo saaray."
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"\n\n(Enter text to convert it to a clean, noise-free voice. This model automatically removes background noise from the generated audio.)"
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),
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examples=[
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["Sidee tahay saaxiib? Maanta waa maalin qurux badan."],
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["Tani waa tijaabo si loo maqlo tayada codka oo saafi ah."],
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["Waan ku faraxsanahay inaan idinla hadlo maanta."],
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]
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)
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