app.py

import gradio as gr
import librosa
import numpy as np
import torch

from transformers import SpeechT5Processor, SpeechT5ForSpeechToSpeech, SpeechT5HifiGan

checkpoint = "microsoft/speecht5_vc"
processor = SpeechT5Processor.from_pretrained(checkpoint)
model = SpeechT5ForSpeechToSpeech.from_pretrained(checkpoint)
vocoder = SpeechT5HifiGan.from_pretrained("microsoft/speecht5_hifigan")


speaker_embeddings = {
    "BDL": "spkemb/cmu_us_bdl_arctic-wav-arctic_a0009.npy",
    "CLB": "spkemb/cmu_us_clb_arctic-wav-arctic_a0144.npy",
    "RMS": "spkemb/cmu_us_rms_arctic-wav-arctic_b0353.npy",
    "SLT": "spkemb/cmu_us_slt_arctic-wav-arctic_a0508.npy",
}


def process_audio(sampling_rate, waveform):
    # convert from int16 to floating point
    waveform = waveform / 32678.0

    # convert to mono if stereo
    if len(waveform.shape) > 1:
        waveform = librosa.to_mono(waveform.T)

    # resample to 16 kHz if necessary
    if sampling_rate != 16000:
        waveform = librosa.resample(waveform, orig_sr=sampling_rate, target_sr=16000)

    # limit to 30 seconds
    waveform = waveform[:16000*30]

    # make PyTorch tensor
    waveform = torch.tensor(waveform)
    return waveform


def predict(speaker, audio, mic_audio=None):
    # audio = tuple (sample_rate, frames) or (sample_rate, (frames, channels))
    if mic_audio is not None:
        sampling_rate, waveform = mic_audio
    elif audio is not None:
        sampling_rate, waveform = audio
    else:
        return (16000, np.zeros(0).astype(np.int16))

    waveform = process_audio(sampling_rate, waveform)
    inputs = processor(audio=waveform, sampling_rate=16000, return_tensors="pt")

    speaker_embedding = np.load(speaker_embeddings[speaker[:3]])
    speaker_embedding = torch.tensor(speaker_embedding).unsqueeze(0)

    speech = model.generate_speech(inputs["input_values"], speaker_embedding, vocoder=vocoder)

    speech = (speech.numpy() * 32767).astype(np.int16)
    return (16000, speech)


title = "SpeechT5: Voice Conversion"

description = """
The <b>SpeechT5</b> model is pre-trained on text as well as speech inputs, with targets that are also a mix of text and speech.
By pre-training on text and speech at the same time, it learns unified representations for both, resulting in improved modeling capabilities.

SpeechT5 can be fine-tuned for different speech tasks. This space demonstrates the <b>speech-to-speech</b> checkpoint for (American) English
language voice conversion.

See also the <a href="https://huggingface.co/spaces/Matthijs/speecht5-asr-demo">speech recognition (ASR) demo</a>
and the <a href="https://huggingface.co/spaces/Matthijs/speecht5-tts-demo">text-to-speech (TTS) demo</a>.

<b>How to use:</b> Upload an audio file or record using the microphone. The audio is converted to mono and resampled to 16 kHz before
being passed into the model. The output is a mel spectrogram, which is converted to a mono 16 kHz waveform by the HiFi-GAN vocoder.
Because the model always applies random dropout, each attempt will give slightly different results.
"""

article = """
<div style='margin:20px auto;'>

<p>References: <a href="https://arxiv.org/abs/2110.07205">SpeechT5 paper</a> |
<a href="https://github.com/microsoft/SpeechT5/">original GitHub</a> |
<a href="https://huggingface.co/mechanicalsea/speecht5-vc">original weights</a></p>

<pre>
@article{Ao2021SpeechT5,
  title   = {SpeechT5: Unified-Modal Encoder-Decoder Pre-training for Spoken Language Processing},
  author  = {Junyi Ao and Rui Wang and Long Zhou and Chengyi Wang and Shuo Ren and Yu Wu and Shujie Liu and Tom Ko and Qing Li and Yu Zhang and Zhihua Wei and Yao Qian and Jinyu Li and Furu Wei},
  eprint={2110.07205},
  archivePrefix={arXiv},
  primaryClass={eess.AS},
  year={2021}
}
</pre>

<p>Example sound credits:<p>

<ul>
<li>"Hmm, I don't know" from <a href="https://freesound.org/people/InspectorJ/sounds/519189/">InspectorJ</a> (CC BY 4.0 license)
<li>"Henry V" excerpt from <a href="https://freesound.org/people/acclivity/sounds/24096/">acclivity</a> (CC BY-NC 4.0 license)
<li>"You can see it in the eyes" from <a href="https://freesound.org/people/JoyOhJoy/sounds/165348/">JoyOhJoy</a> (CC0 license)
<li>"We yearn for time" from <a href="https://freesound.org/people/Sample_Me/sounds/610529/">Sample_Me</a> (CC0 license)
</ul>

<p>Speaker embeddings were generated from <a href="http://www.festvox.org/cmu_arctic/">CMU ARCTIC</a> using <a href="https://huggingface.co/mechanicalsea/speecht5-vc/blob/main/manifest/utils/prep_cmu_arctic_spkemb.py">this script</a>.</p>

</div>
"""

examples = [
    ["BDL (male)", "examples/yearn_for_time.mp3", None],
    ["CLB (female)", "examples/henry5.mp3", None],
    ["RMS (male)", "examples/see_in_eyes.wav", None],
    ["SLT (female)", "examples/hmm_i_dont_know.wav", None],
]

gr.Interface(
    fn=predict,
    inputs=[
        gr.Radio(label="Speaker", choices=["BDL (male)", "CLB (female)", "RMS (male)", "SLT (female)"], value="BDL (male)"),
        gr.Audio(label="Upload Speech", source="upload", type="numpy"),
        gr.Audio(label="Record Speech", source="microphone", type="numpy"),
    ],
    outputs=[
        gr.Audio(label="Converted Speech", type="numpy"),
    ],
    title=title,
    description=description,
    article=article,
    examples=examples,
    #cache_examples=True,
).launch()