bambara-mt / app.py
Aboubacar OUATTARA - kaira
add audios files
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import concurrent
import os
import tempfile
from typing import Optional, Tuple
import numpy as np
import spaces
from transformers import pipeline
import gradio as gr
import torch
import torchaudio
from resemble_enhance.enhancer.inference import denoise, enhance
from flore200_codes import flores_codes
from tts import BambaraTTS
# Check if CUDA is available
device = "cuda" if torch.cuda.is_available() else "cpu"
# Translation pipeline
translation_model = "oza75/nllb-600M-mt-french-bambara"
translator = pipeline("translation", model=translation_model, max_length=512)
# Text-to-Speech pipeline
tts_model = "oza75/bambara-tts"
tts = BambaraTTS(tts_model)
# Function to translate text to Bambara
@spaces.GPU
def translate_to_bambara(text, src_lang):
translation = translator(text, src_lang=src_lang, tgt_lang="bam_Latn")
return translation[0]['translation_text']
# Function to convert text to speech
@spaces.GPU
def text_to_speech(bambara_text, reference_audio: Optional[Tuple] = None):
if reference_audio is not None:
ref_sr, ref_audio = reference_audio
ref_audio = torch.from_numpy(ref_audio)
# Add a channel dimension if the audio is 1D
if ref_audio.ndim == 1:
ref_audio = ref_audio.unsqueeze(0)
# Save the reference audio to a temporary file if it's not None
with tempfile.NamedTemporaryFile(delete=False, suffix='.wav') as tmp:
torchaudio.save(tmp.name, ref_audio, ref_sr)
tmp_path = tmp.name
# Use the temporary file as the speaker reference
sr, audio = tts.text_to_speech(bambara_text, speaker_reference_wav_path=tmp_path)
# Clean up the temporary file
os.unlink(tmp_path)
else:
# If no reference audio provided, proceed with the default
sr, audio = tts.text_to_speech(bambara_text)
audio = audio.mean(dim=0)
return audio, sr
# Function to enhance speech
# @spaces.GPU
# def enhance_speech(audio_array, sampling_rate, solver, nfe, tau, denoise_before_enhancement):
# solver = solver.lower()
# nfe = int(nfe)
# lambd = 0.9 if denoise_before_enhancement else 0.1
#
# @spaces.GPU(duration=360)
# def denoise_audio():
# try:
# return denoise(audio_array, sampling_rate, device)
# except Exception as e:
# print("> Error while denoising : ", str(e))
# return audio_array, sampling_rate
#
# @spaces.GPU(duration=360)
# def enhance_audio():
# try:
# return enhance(audio_array, sampling_rate, device, nfe=nfe, solver=solver, lambd=lambd, tau=tau)
# except Exception as e:
# print("> Error while enhancement : ", str(e))
# return audio_array, sampling_rate
#
# with concurrent.futures.ThreadPoolExecutor() as executor:
# future_denoise = executor.submit(denoise_audio)
# future_enhance = executor.submit(enhance_audio)
#
# denoised_audio, new_sr1 = future_denoise.result()
# enhanced_audio, new_sr2 = future_enhance.result()
#
# # Convert to numpy and return
# return (new_sr1, denoised_audio.cpu().numpy()), (new_sr2, enhanced_audio.cpu().numpy())
@spaces.GPU
def enhance_speech(audio_array, sampling_rate, solver, nfe, tau, denoise_before_enhancement):
solver = solver.lower()
nfe = int(nfe)
lambd = 0.9 if denoise_before_enhancement else 0.1
denoised_audio, new_sr1 = denoise(audio_array, sampling_rate, device)
enhanced_audio, new_sr2 = enhance(audio_array, sampling_rate, device, nfe=nfe, solver=solver, lambd=lambd, tau=tau)
# Convert to numpy and return
return (new_sr1, denoised_audio.cpu().numpy()), (new_sr2, enhanced_audio.cpu().numpy())
def convert_to_int16(audio_array):
if audio_array.dtype == torch.float32:
# Assuming audio_array values are in the range [-1.0, 1.0]
# Scale to int16 range and convert the datatype
audio_array = (audio_array * 32767).to(torch.int16)
return audio_array
# Define the Gradio interface
def _fn(
src_lang,
text,
reference_audio=None,
solver="Midpoint",
nfe=64,
prior_temp=0.5,
denoise_before_enhancement=False
):
source_lang = flores_codes[src_lang]
# Step 1: Translate the text to Bambara
bambara_text = translate_to_bambara(text, source_lang)
# Step 2: Convert the translated text to speech with reference audio
if reference_audio is not None:
audio_array, sampling_rate = text_to_speech(bambara_text, reference_audio)
else:
audio_array, sampling_rate = text_to_speech(bambara_text)
# Step 3: Enhance the audio
denoised_audio, enhanced_audio = enhance_speech(
audio_array,
sampling_rate,
solver,
nfe,
prior_temp,
denoise_before_enhancement
)
print("Audio Array Shape:", audio_array.shape)
print("Sample Rate:", sampling_rate)
print("Audio Array Dtype:", audio_array.dtype)
print("Max Value in Audio Array:", torch.max(audio_array))
print("Min Value in Audio Array:", torch.min(audio_array))
print("Sampling rate type: ", type(sampling_rate))
print("Denoised sampling rate type: ", type(denoised_audio[0]))
print("Enhanced sampling rate type: ", type(enhanced_audio[0]))
# Return all outputs
return (
bambara_text,
(sampling_rate, convert_to_int16(audio_array).numpy()),
(denoised_audio[0], convert_to_int16(denoised_audio[1])),
(enhanced_audio[0], convert_to_int16(enhanced_audio[1]))
)
def main():
lang_codes = list(flores_codes.keys())
# Build Gradio app
app = gr.Interface(
fn=_fn,
inputs=[
gr.Dropdown(label="Source Language", choices=lang_codes, value='French'),
gr.Textbox(label="Text to Translate", lines=3),
gr.Audio(label="Clone your voice (optional)", type="numpy", format="wav"),
gr.Dropdown(
choices=["Midpoint", "RK4", "Euler"], value="Midpoint",
label="ODE Solver (Midpoint is recommended)"
),
gr.Slider(minimum=1, maximum=128, value=64, step=1, label="Number of Function Evaluations"),
gr.Slider(minimum=0.1, maximum=1, value=0.5, step=0.01, label="Prior Temperature"),
gr.Checkbox(value=False, label="Denoise Before Enhancement")
],
outputs=[
gr.Textbox(label="Translated Text"),
gr.Audio(label="Original TTS Audio"),
gr.Audio(label="Denoised Audio"),
gr.Audio(label="Enhanced Audio")
],
title="Bambara Translation and Text to Speech with Audio Enhancement",
description="Translate text to Bambara and convert it to speech with options to enhance audio quality."
)
app.launch(share=False)
if __name__ == "__main__":
main()