infer_onnx.py

import numpy as np
import onnxruntime

from text import text_to_sequence, sequence_to_text
import torch
import gradio as gr
import soundfile as sf
import tempfile
import yaml
import json
import os

from huggingface_hub import hf_hub_download
from time import perf_counter

DEFAULT_SPEAKER_ID = os.environ.get("DEFAULT_SPEAKER_ID", default="caf_08106")

def intersperse(lst, item):
    result = [item] * (len(lst) * 2 + 1)
    result[1::2] = lst
    return result


def process_text(i: int, text: str, device: torch.device):
    print(f"[{i}] - Input text: {text}")
    x = torch.tensor(
        intersperse(text_to_sequence(text, ["catalan_cleaners"]), 0),
        dtype=torch.long,
        device=device,
    )[None]
    x_lengths = torch.tensor([x.shape[-1]], dtype=torch.long, device=device)
    x_phones = sequence_to_text(x.squeeze(0).tolist())
    print(x_phones)
    return x.numpy(), x_lengths.numpy()

MODEL_PATH_MATCHA_MEL=hf_hub_download(repo_id="BSC-LT/matcha-tts-cat-multispeaker", filename="matcha_multispeaker_cat_opset_15_10_steps_2399.onnx")
MODEL_PATH_MATCHA="matcha_hifigan_multispeaker_cat.onnx"
MODEL_PATH_VOCOS=hf_hub_download(repo_id="BSC-LT/vocos-mel-22khz-cat", filename="mel_spec_22khz_cat.onnx")
CONFIG_PATH=hf_hub_download(repo_id="BSC-LT/vocos-mel-22khz-cat", filename="config.yaml")
SPEAKER_ID_DICT="spk_to_id.json"

sess_options = onnxruntime.SessionOptions()
model_matcha_mel= onnxruntime.InferenceSession(str(MODEL_PATH_MATCHA_MEL), sess_options=sess_options, providers=["CPUExecutionProvider"])
model_vocos = onnxruntime.InferenceSession(str(MODEL_PATH_VOCOS), sess_options=sess_options, providers=["CPUExecutionProvider"])
#model_matcha = onnxruntime.InferenceSession(str(MODEL_PATH_MATCHA), sess_options=sess_options, providers=["CPUExecutionProvider"])

speaker_id_dict = json.load(open(SPEAKER_ID_DICT))
speakers = [sp for sp in speaker_id_dict.keys()]
speakers.sort()

def vocos_inference(mel,denoise):

    with open(CONFIG_PATH, "r") as f:
        config = yaml.safe_load(f)

    params = config["feature_extractor"]["init_args"]
    sample_rate = params["sample_rate"]
    n_fft= params["n_fft"]
    hop_length= params["hop_length"]
    win_length = n_fft

    # ONNX inference
    mag, x, y = model_vocos.run(
        None,
        {
            "mels": mel
        },
    )

    # complex spectrogram from vocos output
    spectrogram = mag * (x + 1j * y)
    window = torch.hann_window(win_length)

    if denoise:
        # Vocoder bias
        mel_rand = torch.zeros_like(torch.tensor(mel))
        mag_bias, x_bias, y_bias = model_vocos.run(
            None,
            {
                "mels": mel_rand.float().numpy()
            },
        )

        # complex spectrogram from vocos output
        spectrogram_bias = mag_bias * (x_bias + 1j * y_bias)

        # Denoising
        spec = torch.view_as_real(torch.tensor(spectrogram))
        # get magnitude of vocos spectrogram
        mag_spec = torch.sqrt(spec.pow(2).sum(-1))

        # get magnitude of bias spectrogram
        spec_bias = torch.view_as_real(torch.tensor(spectrogram_bias))
        mag_spec_bias = torch.sqrt(spec_bias.pow(2).sum(-1))

        # substract 
        strength = 0.0025
        mag_spec_denoised = mag_spec - mag_spec_bias * strength
        mag_spec_denoised = torch.clamp(mag_spec_denoised, 0.0)

        # return to complex spectrogram from magnitude
        angle = torch.atan2(spec[..., -1], spec[..., 0] )
        spectrogram = torch.complex(mag_spec_denoised * torch.cos(angle), mag_spec_denoised * torch.sin(angle))

    # Inverse stft
    pad = (win_length - hop_length) // 2
    spectrogram = torch.tensor(spectrogram)
    B, N, T = spectrogram.shape

    print("Spectrogram synthesized shape", spectrogram.shape)
    # Inverse FFT
    ifft = torch.fft.irfft(spectrogram, n_fft, dim=1, norm="backward")
    ifft = ifft * window[None, :, None]

    # Overlap and Add
    output_size = (T - 1) * hop_length + win_length
    y = torch.nn.functional.fold(
        ifft, output_size=(1, output_size), kernel_size=(1, win_length), stride=(1, hop_length),
    )[:, 0, 0, pad:-pad]

    # Window envelope
    window_sq = window.square().expand(1, T, -1).transpose(1, 2)
    window_envelope = torch.nn.functional.fold(
        window_sq, output_size=(1, output_size), kernel_size=(1, win_length), stride=(1, hop_length),
    ).squeeze()[pad:-pad]

    # Normalize
    assert (window_envelope > 1e-11).all()
    y = y / window_envelope
    
    return y


def tts(text:str, spk_name:str, temperature:float, length_scale:float, denoise:bool):
    spk_id = speaker_id_dict[spk_name]
    sid = np.array([int(spk_id)]) if spk_id is not None else None
    text_matcha , text_lengths = process_text(0,text,"cpu") 

    # MATCHA VOCOS
    inputs = {
        "x": text_matcha,
        "x_lengths": text_lengths,
        "scales": np.array([temperature, length_scale], dtype=np.float32),
        "spks": sid
    }
    mel_t0 = perf_counter()
    # matcha mel inference
    mel, mel_lengths = model_matcha_mel.run(None, inputs)
    mel_infer_secs = perf_counter() - mel_t0
    print("Matcha Mel inference time", mel_infer_secs)

    vocos_t0 = perf_counter()
    # vocos inference
    wavs_vocos = vocos_inference(mel,denoise)
    vocos_infer_secs = perf_counter() - vocos_t0
    print("Vocos inference time", vocos_infer_secs)

    with tempfile.NamedTemporaryFile(suffix=".wav", delete=False, dir="/home/user/app") as fp_matcha_vocos:
        sf.write(fp_matcha_vocos.name, wavs_vocos.squeeze(0), 22050, "PCM_24")

    #MATCHA HIFIGAN

    inputs = {
        "x": text_matcha,
        "x_lengths": text_lengths,
        "scales": np.array([temperature, length_scale], dtype=np.float32),
        "spks": sid
    }
    hifigan_t0 = perf_counter()
    print(f"RTF matcha + vocos { (mel_infer_secs + vocos_infer_secs) / (wavs_vocos.shape[1]/22050) }")
    return fp_matcha_vocos.name

## GUI space

title = """
<div style="text-align: center; max-width: 700px; margin: 0 auto;">
    <div
        style="display: inline-flex; align-items: center; gap: 0.8rem; font-size: 1.75rem;"
    > <h1 style="font-weight: 900; margin-bottom: 7px; line-height: normal;">
        Natural and efficient TTS in Catalan
    </h1> </div>
</div>
 """

description = """
 
🍵 Matcha-TTS, a new approach to non-autoregressive neural TTS, that uses conditional flow matching (similar to rectified flows) to speed up ODE-based speech synthesis

For vocoders we use [Vocos](https://huggingface.co/BSC-LT/vocos-mel-22khz-cat) trained in a catalan set of ~28 hours.

[Matcha](https://huggingface.co/BSC-LT/matcha-tts-cat-onnx) was trained using openslr69 and festcat datasets

"""

about = """
## 📄 About
The TTS test about.

## Samples

<table style="font-size:16px">
  <col width="205">
  <col width="205">
<thead>
<tr>
  <td>Col1</td>
  <td>Col2</td>
  <td>Col3</td>
</tr>
</thead>
<tbody>
<tr>
  <td><audio controls="" preload="none" style="width: 200px">audio not supported<source src="https://samplelib.com/lib/preview/mp3/sample-3s.mp3"></audio></td>
  <td><audio controls="" preload="none" style="width: 200px">audio not supported<source src="https://samplelib.com/lib/preview/mp3/sample-6s.mp3"></audio></td>
  <td><audio controls="" preload="none" style="width: 200px">audio not supported<source src="https://samplelib.com/lib/preview/mp3/sample-9s.mp3"></audio></td>
</tr>
<tr>
   <td><audio controls="" preload="none" style="width: 200px">audio not supported<source src="https://samplelib.com/lib/preview/mp3/sample-3s.mp3"></audio></td>
  <td><audio controls="" preload="none" style="width: 200px">audio not supported<source src="https://samplelib.com/lib/preview/mp3/sample-6s.mp3"></audio></td>
  <td><audio controls="" preload="none" style="width: 200px">audio not supported<source src="https://samplelib.com/lib/preview/mp3/sample-9s.mp3"></audio></td>
</tr>
<tr>
  <td><audio controls="" preload="none" style="width: 200px">audio not supported<source src="https://samplelib.com/lib/preview/mp3/sample-3s.mp3"></audio></td>
  <td><audio controls="" preload="none" style="width: 200px">audio not supported<source src="https://samplelib.com/lib/preview/mp3/sample-6s.mp3"></audio></td>
  <td><audio controls="" preload="none" style="width: 200px">audio not supported<source src="https://samplelib.com/lib/preview/mp3/sample-9s.mp3"></audio></td>
</tr>
</tbody></table>
"""

article = "Training and demo by The Language Technologies Unit from Barcelona Supercomputing Center."

vits2_inference = gr.Interface(
    fn=tts,
    inputs=[
        gr.Textbox(
            value="m'ha costat molt desenvolupar una veu, i ara que la tinc no estaré en silenci.",
            max_lines=1,
            label="Input text",
        ),
        gr.Dropdown(
            choices=speakers,
            label="Speaker id",
            value=DEFAULT_SPEAKER_ID,
            info=f"Models are trained on 47 speakers. You can prompt the model using one of these speaker ids."
        ),
        gr.Slider(
            0.1,
            2.0,
            value=0.667,
            step=0.01,
            label="Temperature",
            info=f"Temperature",
        ),
        gr.Slider(
            0.5,
            2.0,
            value=1.0,
            step=0.01,
            label="Length scale",
            info=f"Controls speech pace, larger values for slower pace and smaller values for faster pace",
        ),
        gr.Checkbox(label="Denoise", info="Removes model bias from vocos", value=True),
    ],
    outputs=[gr.Audio(label="Matcha vocos", interactive=False, type="filepath")]
)

about_article = gr.Markdown(about)

demo = gr.Blocks()

with demo:
    gr.Markdown(title)
    gr.Markdown(description)
    gr.TabbedInterface([vits2_inference, about_article], ["Demo", "About"])
    gr.Markdown(article)

demo.queue(max_size=10)
demo.launch(show_api=False, server_name="0.0.0.0", server_port=7860)