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#run this in your terminal, Olaolu
# Set-ExecutionPolicy Unrestricted -Scope Process
import torch
import gradio as gr
import librosa
import os
import base64
import tempfile
import io
import numpy as np
from dotenv import load_dotenv
from transformers import pipeline
from huggingface_hub import login
from google.cloud import translate_v3
from gradio.routes import mount_gradio_app
from spitch import Spitch
from pydub import AudioSegment
import requests
load_dotenv()
spitch_client = Spitch() #activate spitch
# ===========================
# INITIAL SETUP
# ===========================
# Log in to Hugging Face
hf_token = os.getenv("HUGGINGFACE_TOKEN")
if hf_token:
login(token=hf_token)
else:
print("⚠️ No Hugging Face token found. You cannot access private models.")
# Load and decode Google credentials (Base64 Encoding)
creds_b64 = os.getenv("GOOGLE_APPLICATION_CREDENTIALS_JSON") #get the b64 string
if creds_b64:
creds_json = base64.b64decode(creds_b64).decode("utf-8") #decode to json string
temp_file = tempfile.NamedTemporaryFile(delete=False, suffix=".json") #create a temp file
temp_file.write(creds_json.encode("utf-8")) #write json to this file
temp_file.flush() #write
os.environ["GOOGLE_APPLICATION_CREDENTIALS"] = temp_file.name #update cred
else:
print("⚠️ No GCP creds found.")
# Google Cloud project ID
PROJECT_ID = os.getenv("GOOGLE_CLOUD_PROJECT")
# ===========================
# LOAD ASR MODEL
# ===========================
# use device=0 for GPU if available, otherwise CPU (-1)
device = 0 if torch.cuda.is_available() else -1
try:
asr_yoruba = pipeline("automatic-speech-recognition", model="NCAIR1/Yoruba-ASR", device=device)
except Exception as e:
print(f"⚠️ Could not load Yoruba ASR: {e}")
asr_yoruba = None
# English ASR: default to facebook/wav2vec2-base-960h when EN_ASR_MODEL not provided
EN_ASR_MODEL = os.getenv("EN_ASR_MODEL", "facebook/wav2vec2-base-960h")
try:
asr_english = pipeline("automatic-speech-recognition", model=EN_ASR_MODEL, device=device)
# print(f"✅ English ASR loaded: {EN_ASR_MODEL}")
except Exception as e:
print(f"⚠️ Could not load English ASR ({EN_ASR_MODEL}): {e}")
asr_english = None
print("✅ Done loading models!\n")
# ===========================
# TRANSLATION FUNCTION
# ===========================
def translate_text(text: str, mode: str):
"""Translate text according to mode: 'Yoruba → English' or 'English → Yoruba'."""
if not text:
return ""
if mode == "Yoruba → English":
source = "yo"
target = "en-US"
else:
source = "en"
target = "yo"
try:
google_client = translate_v3.TranslationServiceClient()
parent = f"projects/{PROJECT_ID}/locations/global"
response = google_client.translate_text(
contents=[text],
parent=parent,
mime_type="text/plain",
source_language_code=source,
target_language_code=target,
)
return response.translations[0].translated_text
except Exception as e:
print(f"⚠️ Translation failed: {e}")
return "" # fail gracefully
# ===========================
# Google English ASR
# ===========================
# import os
# from google.cloud.speech_v2 import SpeechClient
# from google.cloud.speech_v2.types import cloud_speech
# def quickstart_v2(audio_file: str) -> cloud_speech.RecognizeResponse:
# """Transcribe an audio file.
# Args:
# audio_file (str): Path to the local audio file to be transcribed.
# Returns:
# cloud_speech.RecognizeResponse: The response from the recognize request, containing
# the transcription results
# """
# # Reads a file as bytes
# with open(audio_file, "rb") as f:
# audio_content = f.read()
# # Instantiates a client
# client = SpeechClient()
# config = cloud_speech.RecognitionConfig(
# auto_decoding_config=cloud_speech.AutoDetectDecodingConfig(),
# language_codes=["en-US"],
# model="long",
# )
# request = cloud_speech.RecognizeRequest(
# recognizer=f"projects/{PROJECT_ID}/locations/global/recognizers/_",
# config=config,
# content=audio_content,
# )
# # Transcribes the audio into text
# response = client.recognize(request=request)
# for result in response.results:
# print(f"Transcript: {result.alternatives[0].transcript}")
# return response
# ===========================
# ASR + translate wrapper
# ===========================
def transcribe_and_translate(file_path, mode: str):
"""Return (transcription, translation) for the given file and mode."""
if not file_path:
return "...", ""
# load and standardize audio
audio, sr = librosa.load(file_path, sr=16000)
# choose ASR pipeline
if mode == "Yoruba → English":
asr_result = asr_yoruba(audio)
else:
asr_result = asr_english(audio)
#get the transcription
transcription = asr_result.get("text", "") if isinstance(asr_result, dict) else str(asr_result)
transcription = transcription.capitalize()
translation = translate_text(transcription, mode) if transcription else ""
return transcription, translation
# # ===========================
# # SPITCH TTS
# # ===========================
# def synthesize_tts_to_array(text: str, language: str = "yo", voice: str = "segun"):
# """Given the translation,
# Use Spitch to produce MP3 bytes and convert
# to (sr, numpy_array) for gr.Audio."""
# if not text:
# return None
# #Invoke Spitch
# resp = spitch_client.speech.generate(text=text, language=language, voice=voice, format="mp3")
# #Get the mp3 bytes
# mp3_bytes = resp.read()
# #Get the audio file
# audio = AudioSegment.from_file(io.BytesIO(mp3_bytes), format="mp3")
# sr = audio.frame_rate
# samples = np.array(audio.get_array_of_samples())
# #TODO
# if audio.channels > 1:
# samples = samples.reshape((-1, audio.channels))
# # normalize integer samples -> float32 in [-1, 1]
# max_val = float(1 << (8 * audio.sample_width - 1))
# samples = samples.astype(np.float32) / max_val
# return (sr, samples)
# ===========================
# YARNGPT TTS
# ===========================
def synthesize_tts_to_array(text: str, language: str = "yo", voice: str = "Femi"):
"""Given the translation,
Use YarnGPT to produce MP3 bytes and convert
to (sr, numpy_array) for gr.Audio."""
YARNGPT_API_URL = "https://yarngpt.ai/api/v1/tts"
YARNGPT_API_KEY = os.getenv("YARNGPT_API_KEY")
if not text or not YARNGPT_API_KEY:
print("⚠️Translation or API key is missing")
return None
headers = {
"Authorization": f"Bearer {YARNGPT_API_KEY}"
}
payload = {
"text": text,
"voice": voice,
}
#Invoke YarnGPT
response = requests.post(YARNGPT_API_URL,
headers=headers,
json=payload,
stream=True)
if response.status_code != 200:
print(f"Error: {response.status_code}")
print(response.json())
return None
#Get the mp3 bytes
mp3_bytes = response.content
# Use io.BytesIO to treat the bytes content as a file in memory
try:
audio = AudioSegment.from_file(io.BytesIO(mp3_bytes), format="mp3")
except Exception as e:
print(f"Error processing audio with pydub: {e}")
return None
# Get the sampling rate
sr = audio.frame_rate
samples = np.array(audio.get_array_of_samples())
#TODO: WHAT DOES THIS DO AND WHY?
if audio.channels > 1:
samples = samples.reshape((-1, audio.channels))
# normalize integer samples -> float32 in [-1, 1]
max_val = float(1 << (8 * audio.sample_width - 1))
samples = samples.astype(np.float32) / max_val
return (sr, samples)
# =========================== OLD
# GRADIO INTERFACE
# ===========================
# with gr.Blocks(title="Olùkọ́ | Learn Yoruba") as app:
# gr.Markdown("# 🇳🇬 Olùkọ́")
# gr.Markdown(
# "Comprehensive Yoruba learning tool!"
# )
# # --- Tab 1: ASR + Translator ---
# # single editable textbox + a microphone recorder placed beside it
# with gr.Row():
# output_transcription = gr.Textbox(
# label="✍️ Speak/Type...",
# interactive=True,
# placeholder="Type here or press the mic to speak..."
# )
# mic_recorder = gr.Audio(
# sources="microphone", #accept microphone input
# type="filepath", #store this in a filepath
# label="🎙️",
# show_label=True
# )
# output_translation = gr.Textbox(label="💬 Translation (English)")
# # When the mic recorder finishes, transcribe and place text into the editable textbox
# mic_recorder.change(
# transcribe_audio,
# inputs=mic_recorder, #input to the transcribe_audio function
# outputs=output_transcription, #output goes into this text box
# )
# # When the user types / changes the textbox, translate
# output_transcription.change(
# translate_text,
# inputs=output_transcription, #input to the translate_text function
# outputs=output_translation, #output of the translate_text function gets stored in outputs
# )
# ===========================
# GRADIO INTERFACE - Tab 1 (updated)
# ===========================
with gr.Blocks(title="Olùkọ́ | Learn Yoruba") as app:
gr.Markdown("# 🇳🇬 Olùkọ́")
gr.Markdown("Comprehensive Yoruba learning tool!")
# direction selector
mode = gr.Radio(
choices=["Yoruba → English", "English → Yoruba"],
value="Yoruba → English",
label="Direction"
)
with gr.Row():
# single editable textbox + microphone next to it
#User input textbox
output_transcription = gr.Textbox(
label="✍️ Speak/Type...",
interactive=True
)
# User input microphone
mic_recorder = gr.Audio(
sources="microphone",
type="filepath",
label="🎙️",
show_label=True
)
#Store translation textbox + TTS model in same row
with gr.Row():
#Translation textbox
output_translation = gr.Textbox(label="💬 Translation")
#Button for TTS. TODO
# tts_button = gr.Button("Play TTS")
#Audio for TTS playback. TODO
# tts_audio = gr.Audio(label="TTS Playback", type="numpy", interactive=False)
#TODO
# def _on_tts_click(text, direction):
# """Generate TTS from the translation textbox (no disk write)
# and return (sr, samples)."""
# if not text:
# return None
# # select language/voice mapping as needed
# if direction == "English → Yoruba":
# lang = "yo"
# voice = "Femi"
# else:
# lang = "en"
# voice = "Mary"
# try:
# result = synthesize_tts_to_array(text, language=lang, voice=voice)
# return result # (sr, numpy_array) or None
# except Exception as e:
# print("TTS generation failed:", e)
# return None
# when the mic finishes: transcribe + translate and populate both boxes
mic_recorder.change(
transcribe_and_translate,
inputs=[mic_recorder, mode],
outputs=[output_transcription, output_translation],
)
# when the user types/edits the transcription box, translate according to mode
output_transcription.change(
translate_text,
inputs=[output_transcription, mode],
outputs=output_translation,
)
#TODO If the TTS_Button is pushed, call the _on_tts_click function
#Send the output audi0 (sr, numpy_array) to the tts_audio block
# tts_button.click(
# _on_tts_click,
# inputs=[output_translation, mode],
# outputs=tts_audio
# )
# ===========================
# APP LAUNCH
# ===========================
# mount_gradio_app(api_app, app, path="/")
if __name__ == "__main__":
app.launch() #server_name="0.0.0.0", server_port=7860)
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