Upload app.py

app.py

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+import os
+import numpy as np
+import torch
+from torch import no_grad, LongTensor
+import argparse
+import commons
+from mel_processing import spectrogram_torch
+import utils
+from models import SynthesizerTrn
+import gradio as gr
+import librosa
+import webbrowser
+
+
+from text import text_to_sequence, _clean_text
+device = "cuda:0" if torch.cuda.is_available() else "cpu"
+language_marks = {
+    "Japanese": "",
+    "日本語": "[JA]",
+    "简体中文": "[ZH]",
+    "English": "[EN]",
+    "Mix": "",
+}
+lang = ['日本語', '简体中文', 'English', 'Mix']
+def get_text(text, hps, is_symbol):
+    text_norm = text_to_sequence(text, hps.symbols, [] if is_symbol else hps.data.text_cleaners)
+    if hps.data.add_blank:
+        text_norm = commons.intersperse(text_norm, 0)
+    text_norm = LongTensor(text_norm)
+    return text_norm
+
+def create_tts_fn(model, hps, speaker_ids):
+    def tts_fn(text, speaker, language, speed):
+        if language is not None:
+            text = language_marks[language] + text + language_marks[language]
+        speaker_id = speaker_ids[speaker]
+        stn_tst = get_text(text, hps, False)
+        with no_grad():
+            x_tst = stn_tst.unsqueeze(0).to(device)
+            x_tst_lengths = LongTensor([stn_tst.size(0)]).to(device)
+            sid = LongTensor([speaker_id]).to(device)
+            audio = model.infer(x_tst, x_tst_lengths, sid=sid, noise_scale=.667, noise_scale_w=0.8,
+                                length_scale=1.0 / speed)[0][0, 0].data.cpu().float().numpy()
+        del stn_tst, x_tst, x_tst_lengths, sid
+        return "Success", (hps.data.sampling_rate, audio)
+
+    return tts_fn
+
+def create_vc_fn(model, hps, speaker_ids):
+    def vc_fn(original_speaker, target_speaker, record_audio, upload_audio):
+        input_audio = record_audio if record_audio is not None else upload_audio
+        if input_audio is None:
+            return "You need to record or upload an audio", None
+        sampling_rate, audio = input_audio
+        original_speaker_id = speaker_ids[original_speaker]
+        target_speaker_id = speaker_ids[target_speaker]
+
+        audio = (audio / np.iinfo(audio.dtype).max).astype(np.float32)
+        if len(audio.shape) > 1:
+            audio = librosa.to_mono(audio.transpose(1, 0))
+        if sampling_rate != hps.data.sampling_rate:
+            audio = librosa.resample(audio, orig_sr=sampling_rate, target_sr=hps.data.sampling_rate)
+        with no_grad():
+            y = torch.FloatTensor(audio)
+            y = y / max(-y.min(), y.max()) / 0.99
+            y = y.to(device)
+            y = y.unsqueeze(0)
+            spec = spectrogram_torch(y, hps.data.filter_length,
+                                     hps.data.sampling_rate, hps.data.hop_length, hps.data.win_length,
+                                     center=False).to(device)
+            spec_lengths = LongTensor([spec.size(-1)]).to(device)
+            sid_src = LongTensor([original_speaker_id]).to(device)
+            sid_tgt = LongTensor([target_speaker_id]).to(device)
+            audio = model.voice_conversion(spec, spec_lengths, sid_src=sid_src, sid_tgt=sid_tgt)[0][
+                0, 0].data.cpu().float().numpy()
+        del y, spec, spec_lengths, sid_src, sid_tgt
+        return "Success", (hps.data.sampling_rate, audio)
+
+    return vc_fn
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser()
+    parser.add_argument("--model_dir", default="./inference/G_latest.pth", help="directory to your fine-tuned model")
+    parser.add_argument("--config_dir", default="./inference/finetune_speaker.json", help="directory to your model config file")
+    parser.add_argument("--share", default=False, help="make link public (used in colab)")
+
+    args = parser.parse_args()
+    hps = utils.get_hparams_from_file(args.config_dir)
+
+
+    net_g = SynthesizerTrn(
+        len(hps.symbols),
+        hps.data.filter_length // 2 + 1,
+        hps.train.segment_size // hps.data.hop_length,
+        n_speakers=hps.data.n_speakers,
+        **hps.model).to(device)
+    _ = net_g.eval()
+
+    _ = utils.load_checkpoint(args.model_dir, net_g, None)
+    speaker_ids = hps.speakers
+    speakers = list(hps.speakers.keys())
+    tts_fn = create_tts_fn(net_g, hps, speaker_ids)
+    vc_fn = create_vc_fn(net_g, hps, speaker_ids)
+    app = gr.Blocks()
+    with app:
+        with gr.Tab("Text-to-Speech"):
+            with gr.Row():
+                with gr.Column():
+                    textbox = gr.TextArea(label="Text",
+                                          placeholder="Type your sentence here",
+                                          value="こんにちわ。", elem_id=f"tts-input")
+                    # select character
+                    char_dropdown = gr.Dropdown(choices=speakers, value=speakers[0], label='character')
+                    language_dropdown = gr.Dropdown(choices=lang, value=lang[0], label='language')
+                    duration_slider = gr.Slider(minimum=0.1, maximum=5, value=1, step=0.1,
+                                                label='速度 Speed')
+                with gr.Column():
+                    text_output = gr.Textbox(label="Message")
+                    audio_output = gr.Audio(label="Output Audio", elem_id="tts-audio")
+                    btn = gr.Button("Generate!")
+                    btn.click(tts_fn,
+                              inputs=[textbox, char_dropdown, language_dropdown, duration_slider,],
+                              outputs=[text_output, audio_output], api_name="btn")
+        with gr.Tab("Voice Conversion"):
+            gr.Markdown("""
+                            录制或上传声音，并选择要转换的音色。
+            """)
+            with gr.Column():
+                record_audio = gr.Audio(label="record your voice", source="microphone")
+                upload_audio = gr.Audio(label="or upload audio here", source="upload")
+                source_speaker = gr.Dropdown(choices=speakers, value=speakers[0], label="source speaker")
+                target_speaker = gr.Dropdown(choices=speakers, value=speakers[0], label="target speaker")
+            with gr.Column():
+                message_box = gr.Textbox(label="Message")
+                converted_audio = gr.Audio(label='converted audio')
+            btn = gr.Button("Convert!")
+            btn.click(vc_fn, inputs=[source_speaker, target_speaker, record_audio, upload_audio],
+                      outputs=[message_box, converted_audio])
+    webbrowser.open("http://127.0.0.1:7860")
+    app.launch(share=args.share)
+
+    
+   
+