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{
'': 'Unfortunately, there is no compatible GPU available to support your training.',
'UVR5': 'UVR5 opened ',
'UVR5': 'UVR5 closed',
'MIT, , 使. <br>, 使. <b>LICENSE</b>.': 'This software is open source under the MIT license. The author does not have any control over the software. Users who use the software and distribute the sounds exported by the software are solely responsible. <br>If you do not agree with this clause, you cannot use or reference any codes and files within the software package. See the root directory <b>Agreement-LICENSE.txt</b> for details.',
'0-': '0-Fech dataset',
'0a-UVR5&': '0a-UVR5 webui (for vocal separation, deecho, dereverb and denoise)',
'UVR5-WebUI': 'Open UVR5-WebUI',
'UVR5': 'UVR5 process output log',
'0b-': '0b-Audio slicer',
'': 'Audio slicer input (file or folder)',
'': 'Audio slicer output folder',
'threshold:': 'Noise gate threshold (loudness below this value will be treated as noise',
'min_length:': 'Minimum length',
'min_interval:': 'Minumum interval for audio cutting',
'hop_size:线': 'hop_size: FO hop size, the smaller the value, the higher the accuracy',
'max_sil_kept:': 'Maximum length for silence to be kept',
'': 'Start audio slicer',
'': 'Stop audio cutting',
'max:': 'Loudness multiplier after normalized',
'alpha_mix:': 'alpha_mix: proportion of normalized audio merged into dataset',
'使': 'CPU threads used for audio slicing',
'': 'Audio slicer output log',
'0c-线ASR': '0c-Chinese ASR tool',
'线ASR': 'Start batch ASR',
'ASR': 'Stop ASR task',
'ASR(only)': 'Batch ASR (Chinese only) input folder',
'ASR': 'ASR output log',
'0d-': '0d-Speech to text proofreading tool',
'WebUI': 'Open labelling WebUI',
'': 'path to proofreading text file',
'': 'Proofreading tool output log',
'1-GPT-SoVITS-TTS': '1-GPT-SOVITS-TTS',
'*/': '*Experiment/model name',
'': 'GPU Information',
'SoVITS-G': 'Pretrained SoVITS-G model path',
'SoVITS-D': 'Pretrained SoVITS-D model path',
'GPT': 'Pretrained GPT model path',
'1A-': '1A-Dataset formatting',
'logs/23456': 'output folder (logs/{experiment name}) should have files and folders starts with 23456.',
'*': '*Text labelling file',
'*': '*Audio dataset folder',
' list': 'Training the file name corresponding to the waveform of the waveform in the List file of the audio file',
'1Aa-': '1Aa-Text',
'GPU-': 'GPU number is separated by -, each GPU will run one process ',
'BERT': ' Pretrained BERT model path',
'': 'Start speech-to-text',
'': 'Stop speech-to-text',
'': 'Text processing output',
'1Ab-SSL': '1Ab-SSL self-supervised feature extraction',
'SSL': 'Pretrained SSL model path',
'SSL': 'Start SSL extracting',
'SSL': 'Stop SSL extraction',
'SSL': 'SSL output log',
'1Ac-token': '1Ac-semantics token extraction',
'token': 'Start semantics token extraction',
'token': 'Stop semantics token extraction',
'token': 'Sematics token extraction output log',
'1Aabc-': '1Aabc-One-click formatting',
'': 'Start one-click formatting',
'': 'Stop one-click formatting',
'': 'One-click formatting output',
'1B-': '1B-Fine-tuned training',
'1Ba-SoVITSSoVITS_weights': '1Ba-SoVITS training. The model is located in SoVITS_weights.',
'batch_size': 'Batch size per GPU:',
'total_epoch': 'Total epochs, do not increase to a value that is too high',
'': 'Text model learning rate weighting',
'save_every_epoch': 'Save frequency (save_every_epoch):',
'ckpt': "Save only the latest '.ckpt' file to save disk space:",
'weights': "Save a small final model to the 'weights' folder at each save point:",
'SoVITS': 'Start SoVITS training',
'SoVITS': 'Stop SoVITS training',
'SoVITS': 'SoVITS training output log',
'1Bb-GPTGPT_weights': '1Bb-GPT training. The model is located in GPT_weights.',
'total_epoch': 'Total training epochs (total_epoch):',
'GPT': 'Start GPT training',
'GPT': 'Stop GPT training',
'GPT': 'GPT training output log',
'1C-': '1C-inference',
'SoVITS_weightsGPT_weights5Zero Shot TTS': 'Choose the models from SoVITS_weights and GPT_weights. The default one is a pretrain, so you can experience zero shot TTS.',
'*GPT': '*GPT models list',
'*SoVITS': '*SoVITS models list',
'GPU,1': 'GPU number, can only input ONE integer',
'': 'refreshing model paths',
'TTSWebUI': 'Open TTS inference WEBUI',
'TTSWebUI': 'TTS inference webui output log',
'2-GPT-SoVITS-': '2-GPT-SoVITS-Voice Changer',
'': 'In construction, please wait',
'TTS': 'TTS inference process is opened',
'TTS': 'TTS inference process closed',
'WebUI': 'proofreading tool webui is opened',
'WebUI': 'proofreading tool webui is closed',
'MIT, , 使. , 使. LICENSE.': 'This software is under MIT licence. The author does not have any control for this software. Users are solely reponsible for all voices thats being converted and/or distributed. If you disagree with this Terms and Conditions, you cannot use or cite any files or code in this file. Please check LICENSE. for more info.',
'*': '*Please upload and fill reference information',
'*': '*Please fill the text that needs inference',
'ASR%s': 'ASR training started: %s',
'GPT': 'Finished GPT training',
'GPT%s': 'GPT training started: %s',
'SSL': 'SSL extracting',
'SSL': 'SSL extraction finished',
'SoVITS': 'SoVITS training finished',
'SoVITS%s': 'SoVITS training started%s',
'': 'An error has occured during One-click formatting',
'': 'Finished one-click formatting',
'': 'Chinese',
'50': 'Cut per 50 characters',
'': 'Cut per 5 sentences',
'': 'Text after sliced',
'': 'Slicing audio',
'': 'finished audio slicing',
'': 'Text for reference audio',
'': 'Language for reference audio',
'': 'Start inference',
'': 'Mixed languages input will be supported soon.',
'ASR': ' An ASR task is already in progress, please stop before starting the next task',
'GPT': 'A GPT training task is already in progress, please stop before starting the next task',
'SSL': 'A SSL extraction task is already in progress, please stop before starting the next task',
'SoVITS': 'A SoVITS training task is already in progress, please stop before starting the next task',
'': 'An ASR task is already in progress, please stop before starting the next task',
'': 'An audio slicing task is already in progress, please stop before starting the next task',
'': 'A TTS proofreading task is already in progress, please stop before starting the next task',
'token': 'A semantics token extraction task is already in progress, please stop before starting the next task',
'ASR': 'ASR task has been stopped',
'GPT': 'GPT training has been stopped',
'SoVITS': 'SoVITS training has been stopped',
'1a': 'All 1a tasks has been stopped',
'1b': 'All 1b tasks has been stopped',
'': 'All one-clicking formatting tasks has been stopped',
'': 'All audio slicing tasks has been stopped',
'token': 'All semantics token tasks has been stopped',
'': '',
'': 'Text slicer tool, since there will be issues when infering long texts, so it is advised to cut first. When infering, it will infer respectively then combined together.',
'': 'Text processing',
'': 'Finished text processing',
'': 'Japanese',
'': 'English',
'token': 'Semantics token extracting',
'token': 'Finished semantics token extraction',
'': 'Please upload reference audio',
'': 'No input file or directory',
'': 'Input directory exists, but it is not a file or a folder',
'': 'Inference Result',
'1a-done': 'Progress1a-done',
'1a-done, 1b-ing': 'Progress1a-done, 1b-ing',
'1a-ing': 'Progress1a-ing',
'1a1b-done': 'Progress1a1b-done',
'1a1b-done, 1cing': 'Progress1a1b-done, 1cing',
'all-done': 'Progressall-done',
'': 'Inference text that needs to be sliced',
'': 'Inference text',
'': 'Inference text language',
'>=3使harvest使使': 'If >=3: apply median filtering to the harvested pitch results. The value represents the filter radius and can reduce breathiness.',
'A': 'Weight (w) for Model A:',
'A': 'Path to Model A:',
'B': 'Path to Model B:',
'E:\\+\\\\src': 'C:\\Users\\Desktop\\src',
'F0线, , , F0': 'F0 curve file (optional). One pitch per line. Replaces the default F0 and pitch modulation:',
'Index Rate': 'Index Rate',
'Onnx': 'Export Onnx',
'Onnx': 'Onnx Export Path:',
'RVC': 'RVC Model Path:',
'ckpt': 'ckpt Processing',
'harvest': 'Number of CPU processes used for harvest pitch algorithm',
'index': 'index',
'pth': 'pth',
'rmvpe-使,0-0-1使0211': "Enter the GPU index(es) separated by '-', e.g., 0-0-1 to use 2 processes in GPU0 and 1 process in GPU1",
'step1: . logs, , , , , . ': "Step 1: Fill in the experimental configuration. Experimental data is stored in the 'logs' folder, with each experiment having a separate folder. Manually enter the experiment name path, which contains the experimental configuration, logs, and trained model files.",
'step1:': 'Step 1: Processing data',
'step2:&': 'step2:Pitch extraction & feature extraction',
'step2a: , 2wav; . ': 'Step 2a: Automatically traverse all files in the training folder that can be decoded into audio and perform slice normalization. Generates 2 wav folders in the experiment directory. Currently, only single-singer/speaker training is supported.',
'step2b: 使CPU(), 使GPU()': 'Step 2b: Use CPU to extract pitch (if the model has pitch), use GPU to extract features (select GPU index):',
'step3: , ': 'Step 3: Fill in the training settings and start training the model and index',
'step3a:': 'Step 3a: Model training started',
'': 'One-click training',
', , ': 'Multiple audio files can also be imported. If a folder path exists, this input is ignored.',
' 使UVR5 <br> E:\\codes\\py39\\vits_vc_gpu\\() <br> <br>1HP5HP2HP3HP3HP2 <br>2HP5 <br> 3by FoxJoy<br>\u2003\u2003(1)MDX-Net(onnx_dereverb):<br>&emsp;(234)DeEcho:AggressiveNormalDeReverb<br>/<br>1DeEcho-DeReverb2DeEcho2<br>2MDX-Net-Dereverb<br>3MDX-NetDeEcho-Aggressive': 'Batch processing for vocal accompaniment separation using the UVR5 model.<br>Example of a valid folder path format: D:\\path\\to\\input\\folder (copy it from the file manager address bar).<br>The model is divided into three categories:<br>1. Preserve vocals: Choose this option for audio without harmonies. It preserves vocals better than HP5. It includes two built-in models: HP2 and HP3. HP3 may slightly leak accompaniment but preserves vocals slightly better than HP2.<br>2. Preserve main vocals only: Choose this option for audio with harmonies. It may weaken the main vocals. It includes one built-in model: HP5.<br>3. De-reverb and de-delay models (by FoxJoy):<br>\u2003\u2003(1) MDX-Net: The best choice for stereo reverb removal but cannot remove mono reverb;<br>&emsp;(234) DeEcho: Removes delay effects. Aggressive mode removes more thoroughly than Normal mode. DeReverb additionally removes reverb and can remove mono reverb, but not very effectively for heavily reverberated high-frequency content.<br>De-reverb/de-delay notes:<br>1. The processing time for the DeEcho-DeReverb model is approximately twice as long as the other two DeEcho models.<br>2. The MDX-Net-Dereverb model is quite slow.<br>3. The recommended cleanest configuration is to apply MDX-Net first and then DeEcho-Aggressive.',
'-使, 0-1-2 使012': "Enter the GPU index(es) separated by '-', e.g., 0-1-2 to use GPU 0, 1, and 2:",
'&&': 'Vocals/Accompaniment Separation & Reverberation Removal',
'使': '使',
'使': '使',
'': 'Save name:',
', ': 'Save file name (default: same as the source file):',
'': 'Saved model name (without extension):',
'artifact0.5': 'Protect voiceless consonants and breath sounds to prevent artifacts such as tearing in electronic music. Set to 0.5 to disable. Decrease the value to increase protection, but it may reduce indexing accuracy:',
'': 'Modify',
'(weights)': "Modify model information (only supported for small model files extracted from the 'weights' folder)",
'': 'Stop audio conversion',
'': 'All processes have been completed!',
'': 'Refresh voice list and index path',
'': 'Load model',
'D': 'Load pre-trained base model D path:',
'G': 'Load pre-trained base model G path:',
'': 'Single Inference',
'': 'Unload voice to save GPU memory:',
'(, , 12-12)': 'Transpose (integer, number of semitones, raise by an octave: 12, lower by an octave: -12):',
'0': 'Resample the output audio in post-processing to the final sample rate. Set to 0 for no resampling:',
'': 'No',
'': '',
'': 'Response threshold',
'': 'loudness factor',
'': 'Process data',
'Onnx': 'Export Onnx Model',
'': 'Export file format',
'': 'FAQ (Frequently Asked Questions)',
'': 'General settings',
'': 'Start audio conversion',
'': 'Performance settings',
'': 'Batch Inference',
', , , (opt). ': "Batch conversion. Enter the folder containing the audio files to be converted or upload multiple audio files. The converted audio will be output in the specified folder (default: 'opt').",
'': 'Specify the output folder for vocals:',
'': 'Specify output folder:',
'': 'Specify the output folder for accompaniment:',
'(ms):': 'Inference time (ms):',
'': 'Inferencing voice:',
'': 'Extract',
'使CPU': 'Number of CPU processes used for pitch extraction and data processing:',
'': 'Yes',
'. 10min, ': 'Cache all training sets to GPU memory. Caching small datasets (less than 10 minutes) can speed up training, but caching large datasets will consume a lot of GPU memory and may not provide much speed improvement:',
'': 'View',
'(weights)': "View model information (only supported for small model files extracted from the 'weights' folder)",
'': 'Search feature ratio (controls accent strength, too high has artifacting):',
'': 'Model',
'': 'Model Inference',
'(logs),,': "Model extraction (enter the path of the large file model under the 'logs' folder). This is useful if you want to stop training halfway and manually extract and save a small model file, or if you want to test an intermediate model:",
'': 'Whether the model has pitch guidance:',
'(, )': 'Whether the model has pitch guidance (required for singing, optional for speech):',
',10': 'Whether the model has pitch guidance (1: yes, 0: no):',
'': 'Model architecture version:',
', ': 'Model fusion, can be used to test timbre fusion',
'': 'Path to Model:',
'': 'Fade length',
'': 'Version',
'': 'Feature extraction',
',使': 'Path to the feature index file. Leave blank to use the selected result from the dropdown:',
'+12key, -12key, . ': 'Recommended +12 key for male to female conversion, and -12 key for female to male conversion. If the sound range goes too far and the voice is distorted, you can also adjust it to the appropriate range by yourself.',
'': 'Target sample rate:',
'(ms):': 'Algorithmic delays(ms):',
'index,(dropdown)': 'Auto-detect index path and select from the dropdown:',
'': 'Fusion',
'': 'Model information to be modified:',
'': 'Model information to be placed:',
'': 'Train',
'': 'Train model',
'': 'Train feature index',
', train.log': "Training complete. You can check the training logs in the console or the 'train.log' file under the experiment folder.",
'id': 'Please specify the speaker/singer ID:',
'index': 'Please choose the .index file',
'pth': 'Please choose the .pth file',
'id': 'Select Speaker/Singer ID:',
'': 'Convert',
'': 'Enter the experiment name:',
'': 'Enter the path of the audio folder to be processed:',
'()': 'Enter the path of the audio folder to be processed (copy it from the address bar of the file manager):',
'()': 'Enter the path of the audio file to be processed (default is the correct format example):',
'1使': 'Adjust the volume envelope scaling. Closer to 0, the more it mimicks the volume of the original vocals. Can help mask noise and make volume sound more natural when set relatively low. Closer to 1 will be more of a consistently loud volume:',
'': 'Input voice monitor',
'': 'Enter the path of the training folder:',
'': 'Input device',
'': 'Input noise reduction',
'': 'Output information',
'': 'Output converted voice',
'': 'Output device',
'': 'Output noise reduction',
'(,)': 'Export audio (click on the three dots in the lower right corner to download)',
'.index': 'Select the .index file',
'.pth': 'Select the .pth file',
',pm,harvest,crepeGPU': ',pm,harvest,crepeGPU',
',pm,harvest,crepeGPU,rmvpeGPU': "Select the pitch extraction algorithm ('pm': faster extraction but lower-quality speech; 'harvest': better bass but extremely slow; 'crepe': better quality but GPU intensive), 'rmvpe': best quality, and little GPU requirement",
':pm,CPUdio,harvest,rmvpeCPU/GPU': "Select the pitch extraction algorithm: when extracting singing, you can use 'pm' to speed up. For high-quality speech with fast performance, but worse CPU usage, you can use 'dio'. 'harvest' results in better quality but is slower. 'rmvpe' has the best results and consumes less CPU/GPU",
':': ':',
'': 'Sample length',
'': 'Reload device list',
'': 'Pitch settings',
'(使)': 'Audio device (please use the same type of driver)',
'': 'pitch detection algorithm',
'': 'Extra inference time'
}