Yay! I've been waiting on this one. Thank you so much.

Thanks for listening to us and making this video!

Great help for figuring out all these little details you just have to know somehow. Tnx!

That's great you brought this tutorial for the windows community. I personally use linux to train my models, but it's awesome you are making an effort to make the windows open voice community stronger.

Really good video man. Well explained and researched. Thanks a lot

Thank you very much for your videos. I almst never subscribe but I was so thankful for these that I've been liking every one and I did subscribe. :)

nice! giving Windows some love :D

Great and Unique Videos Always, Thank you for your time and efforts.

Thanks for sharing, Thorsten! Got yourself a new subscriber (y)

Thank you so much for this video, really helpful!

Came here looking some information on Coqui as I'm looking to do a voice clone for voice over work. Fantastic job.

Thank you so much!

Thank you for this tutorial and your entire audio series. I once started with Turtoise, which was too slow for me. Then I found coqui and your public voice model, which is also really good and understandable and with the factor 0.41 is also super fast for me. For my use case, however, still too funny pronunciations of proper names. Through this video I could finally create my own voice model that is completely adapted to the requirements of telling stories. It still sounds a bit shaky here and there and has just 100k steps (with increasing audio material), but is already on the way to improvement. Due to recording conditions and my unfortunately not so great narrator voice. I even come to a loss of 26-36%. So here can still be properly readjusted. For all who are interested in the Sats, if they also want to do something like that: Specs: RTX 2070, I7-10900k, Samsung Evo 970 Steptime: 0.5-0.6 Batchsize (you can go higher): 20 Checkpoint_steps: 1000 (just because i am lazy and train it in the middle of some idle periods, school work etc., so i don't have to wait for 10000) Audio dataset: Specs: HyperX idk (the rgb one) with pop filter, relative big room Here I can't make a statement like this and if you start with the "Total" you will get faster results. I trained in steps with increasing audio files: 0-5k: about 230 files ~ 0.4h 5-10k: about 350 files ~ 0.6h 10-30k: about 500 files ~ 1h 30-60k: about 800 files ~ 1.6h 60k-100k: about 1200 files ~ 2h Current total: 1200 ~ 2h Milestones: from 10k: First beginnings to understand not only noise but (not understandable) from 20k: First word recognizable without knowing text from 30-40k: Understandable text (but not nearly speech) from 80k: It's okay :) * Please note, however, that I used as input books and book excerpts with many proper names and denglish (German with some English words in books). This makes the training process slower in any case and generally worse (but in the trained areas, proper names, very good). Recording: For the recordings I wrote a Python script that automatically splits the text of a text file into sentences (ignoring sentences below 5 words) and outputs them. Then the recording was automatically started and stopped as soon as one second the sound was below 50DB. Then this audio was trimmed so that front and back everything is dropped (below 50DB to garantee a instant speech) and filled with 50ms silence. Then nomalized and saved in ljspeech format. Delete function included

Your content is amazing, really useful. Thx.

this very well done explained.. Thank you Thorston-Voice this video helps me to continue my hobby and research.

Thank you for this

i subscibed 1st video great teacher!!!!!!!!

I love your knowledge man

I spent ages trying to get this to work and finally ended up installing wsl which made the setup work. You should make a video on how to create your own dataset for training! Liebe Grüße aus den USA!

A lot of thanks man !

Thanks a lot!!

Nice thank youu

thank you for your video , it's great worker

Vielen Dank für deine Videos, Thorsten! Du hast meinen Subscribe sicher! Liebe Grüße aus Taiwan :)

I love this if for no other reason it helps me learn German dialects.

Mal wieder klasse Video. Gibt es ein ATI Äquivalent?

Du bist mein Held.

I subscribed although I could only watch for a few mins because of some health problems I’m having nowadays. If possible I Would like a cool tutorial or explanation on ways to do this without downloading anything new to my computer or going through a long process, like maybe if it’s possible to do this 100% online then that would be awesome! Since technology is improving so fast nowadays I’m sure there’s some sites that have to exist where we can do this online right..

Okay 👍🏻👍🏻

Thanks for the video. Also can you make a video on how to run tortoise tts locally on your computer.

Thanks for the tutorial. Its really helpful. Can you also make a tutorial on how can we make use of coqui TTS service to fine-tune yourTTS for low resource language with better quality. That would be really helpful. Thanks and keep inspiring :)

Hi Thorsten, thanks for this awesome tutorial which worked perfectly on my machine. However, I trained my model and it's great but not perfect. Is there an option to continue training with this model instead of training a new one (which would take ages just to get to the point where i am now)? I am relatively new to python, so I am not sure if I just have to modify the training script a little or if there is a command somewhere which does this, or if it's just not possible. If you could give me a pointer that would be great!

Great tutorial! Thank you for all the details! I have a question though about the training process and dataset. I used 102 samples for my dataset. In order to record them I used Audacity with default recordings settings (mono, 44100 Hz, 32-bit float). For the recipe file, I used the one you show in your video (named something like a "youtube recipe"). After 1000 Epochs I checked the results by synthesizing some words and sentences using tts-server. It was sounding very slow, not normal. While checking the congif,json file I found out that the sample rate in was set to 22050. After I changed it to 44100 and restarted the tts-sever voice was sounding closer to mine, but the quality is still really bad. Could the fact that all the samples were recorded at 44100 Hz affect the whole training since the default saple_rate in that config.json file is 22050 or it is irrelevant and I just need to train it more? Or do I need to start over using samples recorded with 22050 Hz frequency?

Hello, thanks so much for the video. I'm in the process of training a custom VITS TTS model using a dataset that I've created. Around the 200,000-step mark, the average loss on my trainEpochstats/avg_loss_1 is creeping up . My dataset is fairly small, approximately 1 hour in length, but it does have good coverage of phonemes. When I tested the audio, it had the correct voice quality but the speech was nonsensical. Should I halt the training to expand my dataset, or is it typical for models to require more training steps to produce meaningful audio?

Hello, Mister @Thorsten, I wanted to know how you do the training a thousand times, and yet the sound does not sound clear, but when I use your voice through the tts-server, the sound appears very clear .... How did you train your voice? (which is on the server) and Thank you for this great effort.

I would have appreciated you breaking down how the audio samples should be formatted, maybe a bit more explanation of the code and also torch audio does not install along with torch either.

Which graphic card do you use pls. Thanks for the info.

Hi Thorsten, thanks for putting the video together, when I try run my version of your train_vits_.py script, I get an error saying ModuleNotFoundError: No module named 'TTS.tts.configs.shared_configs - any pointers (I tried to add the project path to my system environment variable, but no luck)

Hello! Thanks for the tutorial! Just finished teaching. My bot can't string letters into words at all. I would like to ask you what scale the dataset should be, and is it possible to speed up the training with google collab?

Hi, nice video ! Could you tell me what you think of the new arduino for speech recognition ? -> nicla voice

hello! I just wanted to know hoy many audio files do I need to clon a voice, since i just recorded like 50 wavs files but when I start the trainer the script fails since "there is no sample left"

Hello,i just subscribe to your channel and i have one question:does this work with foreign languages or only english?

awesome tutorial, thank you... unfortunately, it keeps getting interrupted with a multiprocess error before the last step, I'm looking for a solution to solve the error. If others have succeeded, and I see in the video that it works for you, maybe it will work for me too. :) Could there be a difference between Windows that could cause this error?

hello Thorsten: I try to figure it out by myself follow the step, but it doesn't work in some how, can i make appointment with you for about half an hour, so that you can give me some guidance?

cool video, can you do this with a docker setup, sans windows?

I started training the model, and after 8 hours, only 2 epochs were completed. Is this normal and do I need to complete all 1000?

hey, I am trying to train my model on my language(kazakh) by your tutrotial. it's been over 1 day since it training, but I am getting some weird noises of speakers, I didn't see that you change or add any symbols, so did I. Do I need to add alphabet of my language?

Hi! Everything works fine, thanx! Except that it refuses to handle accented Hungarian characters (éáűőúöüóí). Does it need to be converted somewhere to handle these letters as well? For sentences without an accented character, it is perfect.

How do I fix the freeze issue? I can't find anything about it other than the resource you provided (bug) was 'closed' with the authors comment being 'we don't support windows' when you've clearly done it on windows! I've spent a lot of time on this and would like to figure it, and help would be appreciated.

Is it possbile to combine two voices? And what sample rate should I use for the dataset?

I would like to training new model tts for new language. Is this the same way to to that? Can you give me some advice it please.. it's really help me

confused where exactly did you put your voice file for training ?

I need help!Inside the folder TTS - training there are some archives as you show in the video, how were these archives found there? How do I put it exactly the same in the folder TTS - Training I made?and when i change directory and enter in the TTS - Training folder and type the python command nothing happens.Please could you help me on that? :(

great tutorial but i am trying to replace my microsoft voices with my cloned voice is this doable?

So i'm getting as far as running the "pip install -e ." command before getting errored out with status code 1 something about wheel

I have a question. How can I stop in the middle and learn from that point again?

I follow every step up until 08:33 but when I run `pip install TTS` it tries to install every version of transformers. I would share a screenshot if I could. Never seen a `pip install` go through all the different versions of a package

Can you change the tone of the voice reading text {e.g. excited, sad, etc}?

Hey Thorsten. Kann man coqui so installieren mit allen Models und Funktionen, wie auf der Website, dass man keine Commands mehr eintippen muss und es komplett offline nutzen kann über das User Interface?

Is there a way to stop and resume training? The continue path command does begin the process but it then fails when generating sample sentences.

I want to make a new model of Indonesian language. but in espeak-ng it doesn't support that language. is it still possible to make a new model?

Sir while running last line, error occurres = charmap, codec can't decide bytes. Plz help

Thanks Thorsten for your endless efforts at communicating a complex subject with enthusiasm and passion to people who don't know much about python. I see that you have linked another video about preparing recordings kzfaq.info/get/bejne/ar-Ea7qLucXcZGw.html

Ich habe Python 3.11 installiert. Muss ich das deinstallieren und 3.8 installieren? Wäre voll kacke

i can't get the pip command to work, help!!

how many samples do we need for the trainnig

How long does it take to train a model? lg

how long training until it sounds good?

5:36 is not very clear where did that come from?

Right, how should I go about creating the dataset though?

11:09 Help please im stuck in this step becuase its gave this error: "OSError: [WinError 126] The specified module could not be found. Error loading "cudart64_110.dll" or one of its dependencies."

Coqui Eleutherodactylus a frog from Puerto Rico 🇵🇷

is it possible to train the model to speak in Portuguese?

Is there a non CUDA version?

The process return the error: "PermissionError: [WinError 32] The process cannot access the file because it is being used by another process..." Im used the your code.

Waiting fro Linux to get proper HQ Text to Speech.

Where is TTS-training??

It looks like mining issues

ei gude wie?! ;)

Well something on my end is not working -.-!

Oh, nur betrug clips stellt der Herr rein, intressant, da gibt es viel zu reporten ...

Thank you, this video has helped me get to this point. Can you help with this error, I am stuck here and can't seam to find a solution. I followed your video but when I go to run the trainer i get the following error: (TTS) C:\Users\7danny\Documents\CoquiTTS\TTS>python .\train_vits_win.py Traceback (most recent call last): File ".\train_vits_win.py", line 6, in from TTS.tts.configs.vits_config import VitsConfig File "C:\Users\7danny\Documents\CoquiTTS\TTS\TTS\tts\configs\vits_config.py", line 5, in from TTS.tts.models.vits import VitsArgs, VitsAudioConfig File "C:\Users\7danny\Documents\CoquiTTS\TTS\TTS\tts\models\vits.py", line 38, in from TTS.vocoder.models.hifigan_generator import HifiganGenerator File "C:\Users\7danny\Documents\CoquiTTS\TTS\TTS\vocoder\models\hifigan_generator.py", line 6, in from torch.nn.utils.parametrizations import weight_norm ImportError: cannot import name 'weight_norm' from 'torch.nn.utils.parametrizations' (C:\Users\7danny\Documents\CoquiTTS\TTS\lib\site-packages\torch n\utils\parametrizations.py)

This is an awesome tutorial, thank you for doing all the trial and error that I kept running into. I do have one problem though. I've used your modified training script and only changed the directories, but I'm still getting a permission error: PermissionError: [WinError 32] The process cannot access the file because it is being used by another process: 'D:/TTS/ThorstenTut/ljsAlex01-April-26-2023_05+12PM-0000000\\events.out.tfevents.1682554375.DESKTOP-IUNHJ2B' Is there any workaround for this? It's pointing to one of the files it just generated, which means it's not being used by any other process, so it must be that multithreading problem you mentioned still being an issue somehow.

It tells me that I might need to install an third party phonemizer for the language de.... Where do you get the extra files from that u have installed and cd.. into at about 10:37 ? I