ClearVoice / app.py
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import torch
import soundfile as sf
import gradio as gr
import spaces
from clearvoice import ClearVoice
import os
import random
@spaces.GPU
def fn_clearvoice_se(input_wav, sr):
if sr == "16000 Hz":
myClearVoice = ClearVoice(task='speech_enhancement', model_names=['FRCRN_SE_16K'])
fs = 16000
else:
myClearVoice = ClearVoice(task='speech_enhancement', model_names=['MossFormer2_SE_48K'])
fs = 48000
output_wav_dict = myClearVoice(input_path=input_wav, online_write=False)
if isinstance(output_wav_dict, dict):
key = next(iter(output_wav_dict))
output_wav = output_wav_dict[key]
else:
output_wav = output_wav_dict
sf.write('enhanced.wav', output_wav[0,:], fs)
return 'enhanced.wav'
@spaces.GPU
def fn_clearvoice_ss(input_wav):
myClearVoice = ClearVoice(task='speech_separation', model_names=['MossFormer2_SS_16K'])
output_wav_dict = myClearVoice(input_path=input_wav, online_write=False)
if isinstance(output_wav_dict, dict):
key = next(iter(output_wav_dict))
output_wav_list = output_wav_dict[key]
output_wav_s1 = output_wav_list[0]
output_wav_s2 = output_wav_list[1]
else:
output_wav_list = output_wav_dict
output_wav_s1 = output_wav_list[0]
output_wav_s2 = output_wav_list[1]
sf.write('separated_s1.wav', output_wav_s1[0,:], 16000)
sf.write('separated_s2.wav', output_wav_s2[0,:], 16000)
return "separated_s1.wav", "separated_s2.wav"
def find_mp4_files(directory):
mp4_files = []
# Walk through the directory and its subdirectories
for root, dirs, files in os.walk(directory):
for file in files:
# Check if the file ends with .mp4
if file.endswith(".mp4") and file[:3] == 'est':
mp4_files.append(os.path.join(root, file))
return mp4_files
@spaces.GPU()
def fn_clearvoice_tse(input_video):
myClearVoice = ClearVoice(task='target_speaker_extraction', model_names=['AV_MossFormer2_TSE_16K'])
#output_wav_dict =
print(f'input_video: {input_video}')
myClearVoice(input_path=input_video, online_write=True, output_path='path_to_output_videos_tse')
output_list = find_mp4_files(f'path_to_output_videos_tse/AV_MossFormer2_TSE_16K/{os.path.basename(input_video).split(".")[0]}/')
return output_list
@spaces.GPU
def fn_clearvoice_sr(input_wav, apply_se):
wavname = input_wav.split('/')[-1]
myClearVoice = ClearVoice(task='speech_super_resolution', model_names=['MossFormer2_SR_48K'])
fs = 48000
if apply_se:
new_wavname = wavname.replace('.wav', str(random.randint(0,1000))+'.wav')
myClearVoice_se = ClearVoice(task='speech_enhancement', model_names=['MossFormer2_SE_48K'])
myClearVoice_se(input_path=input_wav, online_write=True, output_path=new_wavname)
input_wav = new_wavname
output_wav_dict = myClearVoice(input_path=input_wav, online_write=False)
if isinstance(output_wav_dict, dict):
key = next(iter(output_wav_dict))
output_wav = output_wav_dict[key]
else:
output_wav = output_wav_dict
sf.write('enhanced_high_res.wav', output_wav[0,:], fs)
return 'enhanced_high_res.wav'
demo = gr.Blocks()
se_demo = gr.Interface(
fn=fn_clearvoice_se,
inputs = [
gr.Audio(label="Input Audio", type="filepath"),
gr.Dropdown(
["16000 Hz", "48000 Hz"], value="16000 Hz", multiselect=False, info="Choose a sampling rate for your output."
),
],
outputs = [
gr.Audio(label="Output Audio", type="filepath"),
],
title = "<a href='https://github.com/modelscope/ClearerVoice-Studio' target='_blank'>ClearVoice<a/>: Speech Enhancement",
description = ("ClearVoice ([Github Repo](https://github.com/modelscope/ClearerVoice-Studio)) is AI-powered and extracts clear speech from background noise for enhanced speech quality. It supports both 16 kHz and 48 kHz audio outputs. "
"To try it, simply upload your audio, or click one of the examples. "),
article = ("<p style='text-align: center'><a href='https://arxiv.org/abs/2206.07293' target='_blank'>FRCRN: Boosting Feature Representation Using Frequency Recurrence for Monaural Speech Enhancement (ICASSP 2022)</a> </p>"
"<p style='text-align: center'><a href='https://arxiv.org/abs/2312.11825' target='_blank'>MossFormer2: Combining Transformer and RNN-Free Recurrent Network for Enhanced Time-Domain Monaural Speech Separation (ICASSP 2024)</a> </p>"),
examples = [
["examples/mandarin_speech_16kHz.wav", "16000 Hz"],
["examples/english_speech_48kHz.wav", "48000 Hz"],
],
cache_examples = True,
)
ss_demo = gr.Interface(
fn=fn_clearvoice_ss,
inputs = [
gr.Audio(label="Input Audio", type="filepath"),
],
outputs = [
gr.Audio(label="Output Audio", type="filepath"),
gr.Audio(label="Output Audio", type="filepath"),
],
title = "<a href='https://github.com/modelscope/ClearerVoice-Studio' target='_blank'>ClearVoice<a/>: Speech Separation",
description = ("ClearVoice ([Github Repo](https://github.com/modelscope/ClearerVoice-Studio)) is powered by AI and separates individual speech from mixed audio. It supports 16 kHz and two output streams. "
"To try it, simply upload your audio, or click one of the examples. "),
article = ("<p style='text-align: center'><a href='https://arxiv.org/abs/2302.11824' target='_blank'>MossFormer: Pushing the Performance Limit of Monaural Speech Separation using Gated Single-Head Transformer with Convolution-Augmented Joint Self-Attentions (ICASSP 2023)</a> </p>"
"<p style='text-align: center'><a href='https://arxiv.org/abs/2312.11825' target='_blank'>MossFormer2: Combining Transformer and RNN-Free Recurrent Network for Enhanced Time-Domain Monaural Speech Separation (ICASSP 2024)</a> </p>"),
examples = [
['examples/female_female_speech.wav'],
['examples/female_male_speech.wav'],
],
cache_examples = True,
)
tse_demo = gr.Interface(
fn=fn_clearvoice_tse,
inputs = [
gr.Video(label="Input Video"),
],
outputs = [
gr.Gallery(label="Output Video List")
],
title = "<a href='https://github.com/modelscope/ClearerVoice-Studio' target='_blank'>ClearVoice<a/>: Audio-Visual Speaker Extraction",
description = ("ClearVoice ([Github Repo](https://github.com/modelscope/ClearerVoice-Studio)) is AI-powered and extracts each speaker's voice from a multi-speaker video using facial recognition. "
"To try it, simply upload your video, or click one of the examples. "),
# article = ("<p style='text-align: center'><a href='https://arxiv.org/abs/2302.11824' target='_blank'>MossFormer: Pushing the Performance Limit of Monaural Speech Separation using Gated Single-Head Transformer with Convolution-Augmented Joint Self-Attentions (ICASSP 2023)</a> | <a href='https://github.com/alibabasglab/MossFormer' target='_blank'>Github Repo</a></p>"
# "<p style='text-align: center'><a href='https://arxiv.org/abs/2312.11825' target='_blank'>MossFormer2: Combining Transformer and RNN-Free Recurrent Network for Enhanced Time-Domain Monaural Speech Separation (ICASSP 2024)</a> | <a href='https://github.com/alibabasglab/MossFormer2' target='_blank'>Github Repo</a></p>"),
examples = [
['examples/001.mp4'],
['examples/002.mp4'],
],
cache_examples = True,
)
sr_demo = gr.Interface(
fn=fn_clearvoice_sr,
inputs = [
gr.Audio(label="Input Audio", type="filepath"),
gr.Checkbox(label="Apply Speech Enhancement", value=True),
],
outputs = [
gr.Audio(label="Output Audio", type="filepath"),
],
title = "<a href='https://github.com/modelscope/ClearerVoice-Studio' target='_blank'>ClearVoice<a/>: Speech Super Resolution",
description = ("ClearVoice ([Github Repo](https://github.com/modelscope/ClearerVoice-Studio)) is AI-powered and transform low-resolution audio (effective sampling rate ≥ 16 kHz) into crystal-clear, high-resolution audio at 48 kHz. It supports most of audio types. "
"To try it, simply upload your audio, or click one of the examples. "),
article = ("<p style='text-align: center'><a href='https://arxiv.org/abs/2206.07293' target='_blank'>FRCRN: Boosting Feature Representation Using Frequency Recurrence for Monaural Speech Enhancement (ICASSP 2022)</a> </p>"
"<p style='text-align: center'><a href='https://arxiv.org/abs/2312.11825' target='_blank'>MossFormer2: Combining Transformer and RNN-Free Recurrent Network for Enhanced Time-Domain Monaural Speech Separation (ICASSP 2024)</a> </p>"
"<p style='text-align: center'><a href='https://arxiv.org/abs/2501.10045' target='_blank'>HiFi-SR: A Unified Generative Transformer-Convolutional Adversarial Network for High-Fidelity Speech Super-Resolution (ICASSP 2025)</a> </p>"),
examples = [
["examples/mandarin_speech_16kHz.wav", True],
["examples/LJSpeech-001-0001-22k.wav", True],
["examples/LibriTTS_986_129388_24k.wav", True],
["examples/english_speech_48kHz.wav", True],
],
cache_examples = True,
)
with demo:
gr.TabbedInterface([se_demo, ss_demo, sr_demo, tse_demo], ["Task 1: Speech Enhancement", "Task 2: Speech Separation", "Task 3: Speech Super Resolution", "Task 4: Audio-Visual Speaker Extraction"])
demo.launch()