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Browse files- app.py +320 -0
- requirements.txt +11 -0
app.py
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| 1 |
+
import gradio as gr
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| 2 |
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import torch
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| 3 |
+
from transformers import AutoTokenizer, AutoModelForCausalLM
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| 4 |
+
from google.cloud import speech, texttospeech
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+
import os
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import tempfile
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import time
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from pydub import AudioSegment # For audio conversion
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+
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# ==============================================================================
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| 11 |
+
# 1. CONFIGURE AND LOAD N-ATLaS MODEL
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+
# ==============================================================================
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+
MODEL_ID = "NCAIR1/N-ATLaS"
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print(f"Loading model: {MODEL_ID}...")
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tokenizer = AutoTokenizer.from_pretrained(MODEL_ID)
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# Load model for local Mac testing (dtype=torch.float16, no quantization)
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model = AutoModelForCausalLM.from_pretrained(
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MODEL_ID,
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dtype=torch.float16,
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device_map="auto",
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)
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print("✅ N-ATLaS Model loaded.")
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# ==============================================================================
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+
# 2. INITIALIZE GOOGLE CLOUD CLIENTS
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# Assumes GOOGLE_APPLICATION_CREDENTIALS is set in your environment
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# ==============================================================================
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try:
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speech_client = speech.SpeechClient()
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tts_client = texttospeech.TextToSpeechClient()
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print("✅ Google Cloud STT/TTS clients initialized.")
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except Exception as e:
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print(f"🛑 CRITICAL: Could not initialize Google Cloud clients. {e}")
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print(" Make sure you have set the GOOGLE_APPLICATION_CREDENTIALS environment variable.")
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exit()
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| 38 |
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| 39 |
+
# ==============================================================================
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| 40 |
+
# 3. HELPER FUNCTIONS (STT AND TTS)
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| 41 |
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# ==============================================================================
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| 42 |
+
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| 43 |
+
def transcribe_audio(audio_filepath: str, language_code: str):
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"""
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| 45 |
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Converts audio to WAV/LINEAR16 format and transcribes using Google Cloud STT.
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+
"""
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| 47 |
+
if not audio_filepath:
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return ""
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| 49 |
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print(f"Loading audio file: {audio_filepath}")
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| 50 |
+
try:
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| 51 |
+
# Load audio using pydub (handles various input formats)
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| 52 |
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audio = AudioSegment.from_file(audio_filepath)
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| 53 |
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print(" -> AudioSegment loaded successfully.")
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| 54 |
+
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| 55 |
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target_sample_rate = 16000
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| 56 |
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target_channels = 1 # Mono
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| 57 |
+
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| 58 |
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# Resample and convert to mono
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| 59 |
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audio = audio.set_frame_rate(target_sample_rate).set_channels(target_channels)
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| 60 |
+
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| 61 |
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# Get raw PCM data (LINEAR16)
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| 62 |
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wav_data = audio.raw_data
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| 63 |
+
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| 64 |
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print(f"Transcribing {len(wav_data)} bytes with language: {language_code} at {target_sample_rate} Hz...")
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| 65 |
+
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| 66 |
+
# Configure Google STT for LINEAR16 (Default model)
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| 67 |
+
recognition_audio = speech.RecognitionAudio(content=wav_data)
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| 68 |
+
recognition_config = speech.RecognitionConfig(
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| 69 |
+
encoding=speech.RecognitionConfig.AudioEncoding.LINEAR16,
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| 70 |
+
sample_rate_hertz=target_sample_rate,
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| 71 |
+
language_code=language_code,
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| 72 |
+
audio_channel_count=target_channels
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| 73 |
+
)
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| 74 |
+
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| 75 |
+
response = speech_client.recognize(config=recognition_config, audio=recognition_audio)
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| 76 |
+
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| 77 |
+
if not response.results:
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| 78 |
+
return "[Could not understand audio]"
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| 79 |
+
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| 80 |
+
transcribed_text = response.results[0].alternatives[0].transcript
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| 81 |
+
print(f" -> Transcribed: {transcribed_text}")
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| 82 |
+
return transcribed_text
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| 83 |
+
|
| 84 |
+
except Exception as e:
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| 85 |
+
print(f" -> 🛑 ERROR during audio processing or transcription: {e}")
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| 86 |
+
return f"[Error processing audio: {e}]"
|
| 87 |
+
finally:
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| 88 |
+
# Clean up the temporary file created by Gradio
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| 89 |
+
if audio_filepath and os.path.exists(audio_filepath):
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| 90 |
+
try:
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| 91 |
+
os.remove(audio_filepath)
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| 92 |
+
print(f" -> Cleaned up temp file: {audio_filepath}")
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| 93 |
+
except OSError as e:
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| 94 |
+
print(f" -> Error deleting temp file {audio_filepath}: {e}")
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| 95 |
+
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| 96 |
+
def synthesize_speech(text, voice_code):
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| 97 |
+
"""Synthesizes speech using Google Cloud TTS with robust voice selection."""
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| 98 |
+
print(f"Synthesizing speech with requested code: {voice_code}...")
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| 99 |
+
synthesis_input = texttospeech.SynthesisInput(text=text)
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| 100 |
+
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| 101 |
+
# --- Robust Voice Selection Logic ---
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| 102 |
+
selected_voice_name = None
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| 103 |
+
selected_ssml_gender = None
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| 104 |
+
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| 105 |
+
# Use high-quality US WaveNet for any English request
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| 106 |
+
if voice_code.startswith("en"):
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| 107 |
+
selected_language_code = "en-US"
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| 108 |
+
selected_voice_name = "en-US-Wavenet-A"
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| 109 |
+
print(f" -> Using high-quality English voice: {selected_voice_name}")
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| 110 |
+
else:
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| 111 |
+
# For non-English (ha, ig, yo), provide the BASE language code
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| 112 |
+
# and request a specific gender. Google should pick a default.
|
| 113 |
+
selected_language_code = voice_code.split('-')[0] # Use 'ha', 'ig', 'yo'
|
| 114 |
+
selected_ssml_gender = texttospeech.SsmlVoiceGender.FEMALE # Ask for a female voice
|
| 115 |
+
print(f" -> Requesting default FEMALE voice for language: {selected_language_code}")
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| 116 |
+
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| 117 |
+
# Set parameters, omitting 'name' if None
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| 118 |
+
voice_params = {"language_code": selected_language_code}
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| 119 |
+
if selected_voice_name:
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| 120 |
+
voice_params["name"] = selected_voice_name
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| 121 |
+
elif selected_ssml_gender:
|
| 122 |
+
voice_params["ssml_gender"] = selected_ssml_gender
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| 123 |
+
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| 124 |
+
voice = texttospeech.VoiceSelectionParams(**voice_params)
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| 125 |
+
# --- End Voice Selection Logic ---
|
| 126 |
+
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| 127 |
+
audio_config = texttospeech.AudioConfig(
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| 128 |
+
audio_encoding=texttospeech.AudioEncoding.MP3
|
| 129 |
+
)
|
| 130 |
+
|
| 131 |
+
# Diagnostic check for non-English voices
|
| 132 |
+
if not voice_code.startswith("en"):
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| 133 |
+
try:
|
| 134 |
+
print(f"--- Listing available voices for language code: {selected_language_code} ---")
|
| 135 |
+
list_voices_response = tts_client.list_voices(language_code=selected_language_code)
|
| 136 |
+
available_voices = [v.name for v in list_voices_response.voices]
|
| 137 |
+
if available_voices:
|
| 138 |
+
print(f"Available voices found: {available_voices}")
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| 139 |
+
else:
|
| 140 |
+
print("No voices found for this language code.")
|
| 141 |
+
except Exception as list_err:
|
| 142 |
+
print(f" -> ERROR trying to list voices: {list_err}")
|
| 143 |
+
|
| 144 |
+
try:
|
| 145 |
+
response = tts_client.synthesize_speech(
|
| 146 |
+
input=synthesis_input, voice=voice, audio_config=audio_config
|
| 147 |
+
)
|
| 148 |
+
with tempfile.NamedTemporaryFile(suffix=".mp3", delete=False) as fp:
|
| 149 |
+
fp.write(response.audio_content)
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| 150 |
+
temp_audio_path = fp.name
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| 151 |
+
print(f" -> Audio saved to: {temp_audio_path}")
|
| 152 |
+
return temp_audio_path
|
| 153 |
+
except Exception as e:
|
| 154 |
+
print(f" -> 🛑 ERROR during speech synthesis: {e}")
|
| 155 |
+
return None
|
| 156 |
+
|
| 157 |
+
# ==============================================================================
|
| 158 |
+
# 4. CORE CHAT FUNCTION (AS A GENERATOR) - DUAL RESPONSE
|
| 159 |
+
# ==============================================================================
|
| 160 |
+
def speech_to_speech_chat(audio_input, history, input_lang, output_voice):
|
| 161 |
+
"""
|
| 162 |
+
Main function for the Gradio app. Handles filepath audio input, uses 'yield',
|
| 163 |
+
and generates BOTH a translation and a conversational reply.
|
| 164 |
+
"""
|
| 165 |
+
|
| 166 |
+
# --- STAGE 0: Get Filepath ---
|
| 167 |
+
user_audio_path = audio_input # Gradio passes the filepath directly
|
| 168 |
+
if user_audio_path is None:
|
| 169 |
+
# Handle case where user clicks submit without recording
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| 170 |
+
yield history, None, None
|
| 171 |
+
return # Stop processing
|
| 172 |
+
print(f"Received audio filepath: {user_audio_path}")
|
| 173 |
+
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| 174 |
+
# ----- STAGE 1: Transcribe User -----
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| 175 |
+
transcribed_text = transcribe_audio(user_audio_path, input_lang) # Pass filepath
|
| 176 |
+
|
| 177 |
+
if transcribed_text is None:
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| 178 |
+
print(" -> 🛑 Transcription function returned None unexpectedly.")
|
| 179 |
+
transcribed_text = "[Error: Transcription failed internally]"
|
| 180 |
+
|
| 181 |
+
history.append((transcribed_text, None))
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| 182 |
+
yield history, None, None # Update UI with transcribed text
|
| 183 |
+
|
| 184 |
+
if transcribed_text.startswith("["):
|
| 185 |
+
return # Stop processing if transcription failed
|
| 186 |
+
|
| 187 |
+
# ----- STAGE 2: Get N-ATLaS Response (RUN 1: CONVERSATION) -----
|
| 188 |
+
print("Generating N-ATLaS response (Run 1: Conversation)...")
|
| 189 |
+
|
| 190 |
+
# Get target language name
|
| 191 |
+
if output_voice.startswith("ha"): lang = "Hausa"
|
| 192 |
+
elif output_voice.startswith("yo"): lang = "Yoruba"
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| 193 |
+
elif output_voice.startswith("ig"): lang = "Igbo"
|
| 194 |
+
else: lang = "Nigerian English"
|
| 195 |
+
|
| 196 |
+
# Create persona prompt for conversation
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| 197 |
+
system_prompt = f"You are a helpful, friendly assistant. Listen to what the user says and respond naturally. You must respond ONLY in {lang}."
|
| 198 |
+
|
| 199 |
+
# Build conversation history
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| 200 |
+
messages = []
|
| 201 |
+
for user_msg, assistant_msg in history:
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| 202 |
+
user_content = str(user_msg) if user_msg is not None else "[empty]"
|
| 203 |
+
messages.append({"role": "user", "content": user_content})
|
| 204 |
+
if assistant_msg:
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| 205 |
+
# Extract just the conversational part from previous turns
|
| 206 |
+
if "**Conversational Reply:**" in str(assistant_msg):
|
| 207 |
+
reply_text = str(assistant_msg).split("---")[0].replace("**Conversational Reply:**\n", "").strip()
|
| 208 |
+
messages.append({"role": "assistant", "content": reply_text})
|
| 209 |
+
else:
|
| 210 |
+
messages.append({"role": "assistant", "content": str(assistant_msg)})
|
| 211 |
+
|
| 212 |
+
# Add the final system prompt
|
| 213 |
+
conversation_messages = messages + [{"role": "system", "content": system_prompt}]
|
| 214 |
+
conversation_prompt = tokenizer.apply_chat_template(conversation_messages, tokenize=False, add_generation_prompt=True)
|
| 215 |
+
|
| 216 |
+
inputs = tokenizer(conversation_prompt, return_tensors="pt").to(model.device)
|
| 217 |
+
input_length = inputs.input_ids.shape[1]
|
| 218 |
+
|
| 219 |
+
outputs = model.generate(
|
| 220 |
+
**inputs, max_new_tokens=256, eos_token_id=tokenizer.eos_token_id,
|
| 221 |
+
pad_token_id=tokenizer.eos_token_id, do_sample=True, temperature=0.7, top_p=0.9
|
| 222 |
+
)
|
| 223 |
+
conversational_text = tokenizer.decode(outputs[0][input_length:], skip_special_tokens=True).strip()
|
| 224 |
+
print(f" -> Conversational Reply: {conversational_text}")
|
| 225 |
+
|
| 226 |
+
# ----- STAGE 3: Get N-ATLaS Response (RUN 2: TRANSLATION) -----
|
| 227 |
+
print("Generating N-ATLaS response (Run 2: Translation)...")
|
| 228 |
+
|
| 229 |
+
translation_system_prompt = f"Translate the following text to {lang}:"
|
| 230 |
+
|
| 231 |
+
translation_messages = [
|
| 232 |
+
{"role": "system", "content": translation_system_prompt},
|
| 233 |
+
{"role": "user", "content": transcribed_text} # Translate only the last user input
|
| 234 |
+
]
|
| 235 |
+
translation_prompt = tokenizer.apply_chat_template(translation_messages, tokenize=False, add_generation_prompt=True)
|
| 236 |
+
|
| 237 |
+
inputs = tokenizer(translation_prompt, return_tensors="pt").to(model.device)
|
| 238 |
+
input_length = inputs.input_ids.shape[1]
|
| 239 |
+
|
| 240 |
+
outputs = model.generate(
|
| 241 |
+
**inputs, max_new_tokens=256, eos_token_id=tokenizer.eos_token_id,
|
| 242 |
+
pad_token_id=tokenizer.eos_token_id, do_sample=False, temperature=0.1, top_p=0.9 # Lower temp for translation
|
| 243 |
+
)
|
| 244 |
+
translation_text = tokenizer.decode(outputs[0][input_length:], skip_special_tokens=True).strip()
|
| 245 |
+
print(f" -> Direct Translation: {translation_text}")
|
| 246 |
+
|
| 247 |
+
# ----- STAGE 4: Synthesize and Format Response -----
|
| 248 |
+
|
| 249 |
+
# Synthesize speech for the CONVERSATIONAL reply only
|
| 250 |
+
bot_audio_path = synthesize_speech(conversational_text, output_voice)
|
| 251 |
+
|
| 252 |
+
# Format a single string for the chatbot UI
|
| 253 |
+
bot_response_string = f"""
|
| 254 |
+
**Conversational Reply:**
|
| 255 |
+
{conversational_text}
|
| 256 |
+
|
| 257 |
+
---
|
| 258 |
+
**Direct Translation:**
|
| 259 |
+
{translation_text}
|
| 260 |
+
"""
|
| 261 |
+
|
| 262 |
+
# Update the history with the user's text and the bot's combined text
|
| 263 |
+
final_user_text = transcribed_text if transcribed_text is not None else "[Error]"
|
| 264 |
+
history[-1] = (final_user_text, bot_response_string)
|
| 265 |
+
|
| 266 |
+
# Yield the final history, the bot's audio, and clear the mic input
|
| 267 |
+
yield history, bot_audio_path, None
|
| 268 |
+
|
| 269 |
+
# ==============================================================================
|
| 270 |
+
# 5. GRADIO UI (using Blocks) - Gradio 3.x compatible
|
| 271 |
+
# ==============================================================================
|
| 272 |
+
with gr.Blocks(theme=gr.themes.Soft(), title="N-ATLaS Voice Test").queue() as iface:
|
| 273 |
+
gr.Markdown("# 🇳🇬 N-ATLaS Multilingual Voice Test")
|
| 274 |
+
gr.Markdown(
|
| 275 |
+
"**Instructions:** Select your spoken language and desired response voice. "
|
| 276 |
+
"Speak into the microphone, then press 'Submit'.\n"
|
| 277 |
+
"**⚠️ IMPORTANT: Response from the N-ATLaS 8B model may take 30-90 seconds locally.**"
|
| 278 |
+
)
|
| 279 |
+
with gr.Row():
|
| 280 |
+
input_lang = gr.Dropdown(
|
| 281 |
+
label="1. Language I am Speaking",
|
| 282 |
+
choices=[
|
| 283 |
+
("American English", "en-US"),
|
| 284 |
+
("Nigerian Pidgin / English", "en-NG"),
|
| 285 |
+
("Hausa", "ha-NG"),
|
| 286 |
+
("Igbo", "ig-NG"),
|
| 287 |
+
("Yoruba", "yo-NG")
|
| 288 |
+
],
|
| 289 |
+
value="en-US" # Default to US English for local testing
|
| 290 |
+
)
|
| 291 |
+
output_voice = gr.Dropdown(
|
| 292 |
+
label="2. Language for Bot to Speak",
|
| 293 |
+
choices=[
|
| 294 |
+
("Nigerian English", "en-NG"),
|
| 295 |
+
("Hausa", "ha-NG"),
|
| 296 |
+
("Igbo", "ig-NG"),
|
| 297 |
+
("Yoruba", "yo-NG")
|
| 298 |
+
],
|
| 299 |
+
value="en-NG"
|
| 300 |
+
)
|
| 301 |
+
chatbot = gr.Chatbot(label="Conversation", height=400)
|
| 302 |
+
mic_input = gr.Audio(
|
| 303 |
+
source="microphone", # Use 'source' (singular) for Gradio 3.x
|
| 304 |
+
type="filepath", # Use 'filepath'
|
| 305 |
+
label="3. Press record and speak"
|
| 306 |
+
)
|
| 307 |
+
bot_audio_output = gr.Audio(
|
| 308 |
+
label="Bot's Spoken Response",
|
| 309 |
+
autoplay=True
|
| 310 |
+
)
|
| 311 |
+
submit_btn = gr.Button("Submit Audio")
|
| 312 |
+
chat_history = gr.State([])
|
| 313 |
+
submit_btn.click(
|
| 314 |
+
fn=speech_to_speech_chat,
|
| 315 |
+
inputs=[mic_input, chat_history, input_lang, output_voice],
|
| 316 |
+
outputs=[chatbot, bot_audio_output, mic_input]
|
| 317 |
+
)
|
| 318 |
+
|
| 319 |
+
print("Launching Gradio interface...")
|
| 320 |
+
iface.launch(share=True) # share=True for public link, remove queue=True
|
requirements.txt
ADDED
|
@@ -0,0 +1,11 @@
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| 1 |
+
gradio==3.50.2
|
| 2 |
+
transformers
|
| 3 |
+
torch
|
| 4 |
+
accelerate
|
| 5 |
+
bitsandbytes
|
| 6 |
+
sentencepiece
|
| 7 |
+
google-cloud-speech
|
| 8 |
+
google-cloud-texttospeech
|
| 9 |
+
ffmpeg-python
|
| 10 |
+
pydub
|
| 11 |
+
pydantic<2
|