FAQ (Frequently Asked Questions)
It is most likely not a FFmpeg issue, but rather an audio path issue;
FFmpeg may encounter an error when reading paths containing special characters like spaces and (), which may cause an FFmpeg error; and when the training set's audio contains Chinese paths, writing it into filelist.txt may cause a utf8 error.
If it displays "Training is done. The program is closed," then the model has been trained successfully, and the subsequent errors are fake;
The lack of an 'added' index file after One-click training may be due to the training set being too large, causing the addition of the index to get stuck; this has been resolved by using batch processing to add the index, which solves the problem of memory overload when adding the index. As a temporary solution, try clicking the "Train Index" button again.
Click “Refresh timbre list” and check again; if still not visible, check if there are any errors during training and send screenshots of the console, web UI, and logs/experiment_name/*.log to the developers for further analysis.
The pth files stored in rvc_root/logs/experiment_name are not meant for sharing or inference, but for storing the experiment checkpoits for reproducibility and further training. The model to be shared should be the 60+MB pth file in the weights folder;
In the future, weights/exp_name.pth and logs/exp_name/added_xxx.index will be merged into a single weights/exp_name.zip file to eliminate the need for manual index input; so share the zip file, not the pth file, unless you want to continue training on a different machine;
Copying/sharing the several hundred MB pth files from the logs folder to the weights folder for forced inference may result in errors such as missing f0, tgt_sr, or other keys. You need to use the ckpt tab at the bottom to manually or automatically (if the information is found in the logs/exp_name), select whether to include pitch infomation and target audio sampling rate options and then extract the smaller model. After extraction, there will be a 60+ MB pth file in the weights folder, and you can refresh the voices to use it.
You may have closed the console (black command line window).
Please disable system LAN proxy/global proxy and then refresh.
Training script:
You can run training in WebUI first, and the command-line versions of dataset preprocessing and training will be displayed in the message window.
Inference script:
https://huggingface.co/lj1995/VoiceConversionWebUI/blob/main/myinfer.py
e.g.
runtime\python.exe myinfer.py 0 "E:\codes\py39\RVC-beta\todo-songs\1111.wav" "E:\codes\py39\logs\mi-test\added_IVF677_Flat_nprobe_7.index" harvest "test.wav" "weights/mi-test.pth" 0.6 cuda:0 True
f0up_key=sys.argv[1]
input_path=sys.argv[2]
index_path=sys.argv[3]
f0method=sys.argv[4]#harvest or pm
opt_path=sys.argv[5]
model_path=sys.argv[6]
index_rate=float(sys.argv[7])
device=sys.argv[8]
is_half=bool(sys.argv[9])
Update:0528v2
https://huggingface.co/lj1995/VoiceConversionWebUI/blob/main/myinfer-v2-0528.py
e.g.
runtime\python.exe myinfer-v2-0528.py 0 "E:\codes\py39\RVC-beta\todo-songs\1111.wav" "E:\codes\py39\test-20230416b\logs\mi-test-v2\aadded_IVF677_Flat_nprobe_1_v2.index" harvest "test_v2.wav" "E:\codes\py39\test-20230416b\weights\mi-test-v2.pth" 0.66 cuda:0 True 3 0 1 0.33
f0up_key=sys.argv[1]
input_path=sys.argv[2]
index_path=sys.argv[3]
f0method=sys.argv[4]#harvest or pm or crepe
opt_path=sys.argv[5]
model_path=sys.argv[6]
index_rate=float(sys.argv[7])
device=sys.argv[8]
is_half=bool(sys.argv[9])
filter_radius=int(sys.argv[10])
resample_sr=int(sys.argv[11])
rms_mix_rate=float(sys.argv[12])
protect=float(sys.argv[13])
There is a small chance that there is a problem with the CUDA configuration or the device is not supported; more likely, there is not enough memory (out of memory).
For training, reduce the batch size (if reducing to 1 is still not enough, you may need to change the graphics card); for inference, adjust the x_pad, x_query, x_center, and x_max settings in the config.py file as needed. 4G or lower memory cards (e.g. 1060(3G) and various 2G cards) can be abandoned, while 4G memory cards still have a chance.
If the training dataset's audio quality is poor and the noise floor is high, 20-30 epochs are sufficient. Setting it too high won't improve the audio quality of your low-quality training set.
If the training set audio quality is high, the noise floor is low, and there is sufficient duration, you can increase it. 200 is acceptable (since training is fast, and if you're able to prepare a high-quality training set, your GPU likely can handle a longer training duration without issue).
A dataset of around 10min to 50min is recommended.
With guaranteed high sound quality and low bottom noise, more can be added if the dataset's timbre is uniform.
For a high-level training set (lean + distinctive tone), 5min to 10min is fine.
There are some people who have trained successfully with 1min to 2min data, but the success is not reproducible by others and is not very informative.
This requires that the training set has a very distinctive timbre (e.g. a high-frequency airy anime girl sound) and the quality of the audio is high;
Data of less than 1min duration has not been successfully attempted so far. This is not recommended.
If the tone quality of the pre-trained model and inference source is higher than that of the training set, they can bring up the tone quality of the inference result, but at the cost of a possible tone bias towards the tone of the underlying model/inference source rather than the tone of the training set, which is generally referred to as "tone leakage".
The index rate is used to reduce/resolve the timbre leakage problem. If the index rate is set to 1, theoretically there is no timbre leakage from the inference source and the timbre quality is more biased towards the training set. If the training set has a lower sound quality than the inference source, then a higher index rate may reduce the sound quality. Turning it down to 0 does not have the effect of using retrieval blending to protect the training set tones.
If the training set has good audio quality and long duration, turn up the total_epoch, when the model itself is less likely to refer to the inferred source and the pretrained underlying model, and there is little "tone leakage", the index_rate is not important and you can even not create/share the index file.
In the config.py file, select the card number after "device cuda:".
The mapping between card number and graphics card can be seen in the graphics card information section of the training tab.
Save via model extraction at the bottom of the ckpt processing tab.
Too many processes and your memory is not enough. You may fix it by:
1、decrease "Number of CPU threads".
2、pre-cut trainset to shorter audio files.
step1: put all wav data to path2
step2: exp_name2+path2 -> process dataset and extract feature
step3: copy the latest G and D file of exp_name1 (your previous experiment) into exp_name2 folder
step4: click "train the model", and it will continue training from the beginning of your previous exp model epoch
OSError: Could not load shared object file: llvmlite.dll
FileNotFoundError: Could not find module lib\site-packages\llvmlite\binding\llvmlite.dll (or one of its dependencies). Try using the full path with constructor syntax.
The issue will happen in windows, install https://aka.ms/vs/17/release/vc_redist.x64.exe and it will be fixed.
Q17：RuntimeError: The expanded size of the tensor (17280) must match the existing size (0) at non-singleton dimension 1. Target sizes: [1, 17280]. Tensor sizes: [0]
Delete the wav files whose size is significantly smaller than others, and that won't happen again. Than click "train the model"and "train the index".
Q18: RuntimeError: The size of tensor a (24) must match the size of tensor b (16) at non-singleton dimension 2
Do not change the sampling rate and then continue training. If it is necessary to change, the exp name should be changed and the model will be trained from scratch. You can also copy the pitch and features (0/1/2/2b folders) extracted last time to accelerate the training process.
