TTS VS piper

lol this marketing technique is getting very old. https://github.com/coqui-ai/TTS is already amazing and open source.

Not sure how relevant this is but note that Coqui TTS (the realistic TTS) has already shut down

https://coqui.ai

I've used coqui.ai's TTS models[0] and library[1] to great success. I was able to get cloned voice to be rendered in about 80% of the audio clip length, and I believe you can also stream the response. Do note the model license for XTTS, it is one they wrote themselves that has some restrictions.

[0] https://huggingface.co/coqui/XTTS-v2

[1] https://github.com/coqui-ai/TTS

My only exposure to Coqui was their text to speech software. If I remember correctly the website was a commercialized service with TTS and probably some other related things. I hope the software work continues in the open.

https://github.com/coqui-ai/TTS

Coqui.ai TTS (https://github.com/coqui-ai/TTS)

"tts_models/multilingual/multi-dataset/xtts_v2" model from https://github.com/coqui-ai/TTS. It gives pretty good results and works with references, so it's pretty easy to change the voice. By the way the source code of the project is open: https://github.com/volotat/Anagnorisis but be ready, the code is pretty raw for now.

A recent update to their GitHub also has a no-code gradio ui to facilitate fine-tuning and inferencing locally. https://github.com/coqui-ai/TTS/releases/tag/v0.21.3

No API token found for 🐸Coqui Studio voices - https://coqui.ai

If you're not already aware, the primary developer of Mimic 3 (and its non-Mimic predecessor Larynx) continued TTS-related development with Larynx and the renamed project Piper: https://github.com/rhasspy/piper

Last year Piper development was supported by Nabu Casa for their "Year of Voice" project for Home Assistant and it sounds like Mike Hansen is going to continue on it with their support this year.

Coqui-ai was a commercial continuation of Mozilla TTS and STT (https://github.com/mozilla/TTS).

At the time (2018-ish), it was really impressive for on-device voice synthesis (with a quality approaching the Google and Azure cloud-based voice synthesis options) and open source, so a lot of people in the FOSS community were hoping it could be used for a privacy-respecting home assistant, Linux speech synthesis that doesn't suck, etc.

After Mozilla abandoned the project, Coqui continued development and had some really impressive one-shot voice cloning, but pivoted to marketing speech synthesis for game developers. They were probably having trouble monetizing it, and it doesn't surprise me that they shut down.

An equivalent project that's still in active development and doing really well is Piper TTS (https://github.com/rhasspy/piper).

There isn't an ElevenLabs app like that, but I think that's the most expedient method, by far.

(details and warning: in-depth, opinionated take, written almost for my own benefit, I've done a lot of work near here recently but haven't had to organize my thoughts until now)

Why? Local inference is hard. You need two things: the clips to voice model (which we have here, but bleeding edge), and text + voice -> speech model.

Text to voice to speech, locally, has excellent prior art for me, in the form of a Raspberry Pi-based ONNX inference library called [Piper](https://github.com/rhasspy/piper). I should just be able to copy that, about an afternoon of work!

Except...when these models are trained, they encode plaintext to model input using a library called eSpeak. eSpeak is basically f(plaintext) => ints representing phonemes. eSpeak is a C library and written in a style I haven't seen in a while and depends on other C libraries. So I end up needing to port like 20K lines of C to Dart...or I could use WASM, but over the last year, I lost the ability to be able to reason through how to get WASM running in Dart, both native and web.

It's a really annoying technical problem: the speech models all use this eSpeak C library to turn plaintext => model input (tokenized phonemes).

Re: ElevenLabs

I had looked into the API months ago and vaguely remembered it was _very_ complete.

I spent the last hour or two playing with it, and reconfirmed that. They have enough API surface that you could build an API that took voice recordings, created a voice, and then did POSTs / socket connection to get audio data from that voice at will.

Only issue is pricing IMHO, $0.18 for 1000 characters. :/ But this is something I feel very comfortable saying wouldn't be _that_ much work to build and open source with a "bring your own API key" type thing. I had forgotten about Eleven Labs till your post, which made me realize there was an actually meaningful and quite moving use case for it.

piper (https://github.com/rhasspy/piper)

I have been using piper-tts and it is GREAT and super lightweight / easy to use. On a 2080 I'm sure you can use the HQ models no worries!

So i found some library and one which is from github and have read.me or good documentation called piper (https://github.com/rhasspy/piper) so apparently this library is for rasbery pi and yes there is TXT function and i need to modify again to make it more simple but my simple project don't need this kind of big complex libary and all i need is what i said before just a function that can output sound from computer using c++ libary.

piper-whistle is a tool to manage voices used with the piper (https://github.com/rhasspy/piper) speech synthesizer. Main motivation was to download and reference models in a structured way. You may browse the docs online at https://think-biq.gitlab.io/piper-whistle/

You may want to try Piper for this case (RPi 4): https://github.com/rhasspy/piper

rhasspy/piper