awesome-ml VS piper

I do something like that for open source:

https://github.com/underlines/awesome-ml

But it lost a bit of traction lately.

It needs re-work for the categories, or better, a tagging system, because these products and libraries can sit in more than one space.

Plus it either needs massive collaboration, or some form of automation (with an LLM and indexer), as I can't keep up with it.

This aera is barely new. Look at how old some of the projects are:

https://github.com/underlines/awesome-ml/blob/master/audio-a...

The thing that changes is the complexity to run it. I was training my wife's voice and my voice for fun and needed 15min of audio and trained on my 3080 for 40 minutes.

Now it's 2 Minutes.

Thanks for your clarifications. I added it to my awesome list:

https://github.com/underlines/awesome-marketing-datascience/...

RAG is very difficult to do right. I am experimenting with various RAG projects from [1]. The main problems are:

- Chunking can interfer with context boundaries

- Content vectors can differ vastly from question vectors, for this you have to use hypothetical embeddings (they generate artificial questions and store them)

- Instead of saving just one embedding per text-chuck you should store various (text chunk, hypothetical embedding questions, meta data)

- RAG will miserably fail with requests like "summarize the whole document"

- to my knowledge, openAI embeddings aren't performing well, use a embedding that is optimized for question answering or information retrieval and supports multi language. Also look into instructor embeddings: https://github.com/embeddings-benchmark/mteb

1 https://github.com/underlines/awesome-marketing-datascience/...

I do the same and with currently with 700+ github stars people seem to like it, but it's still curated/manual, because the hf search API is so limited and I don't have the time to create a scraper.

Added to the list

I keep mine fairly up to date as well, almost daily: https://github.com/underlines/awesome-marketing-datascience/blob/master/README.md

Disclaimer: I am curating LLM-tools on github [1]

A few thoughts:

* allow for custom endpoint URLs, this way people can use open source LLMs with a fake openAI API backend like basaran[2] or llama-api-server[3]

* look into better embedding methods for info-retrieval like InstructorEmbeddings or Document Summary Index

* Don't use a single embedding per content item, use multiple to increase retrieval quality

1 https://github.com/underlines/awesome-marketing-datascience/...

2 https://github.com/hyperonym/basaran

3 https://github.com/iaalm/llama-api-server

Oobabooga isn't a wrapper for llama.cpp, but it can act as such. A usual Oobabooga installation on windows will use a GPTQ wheel (binary) compiled for cuda/windows, or alternatively use llama.cpp's API and act as a GUI. On Linux you had the choice to use the triton or cuda branch for GPTQ, but I don't know if that is still the case. You can also go the route to use virtualized and hardware accelerated WSL2 Ubuntu on Windows and use anything similar to linux. See my guide

Mozilla's browser tts is kind of not bad, just parse and buffer one sentence at a time and it does all right.

For the backend, I've experimented with piper, which has a lot of voices and accents, though it's tricky to buffer and sync long texts.

https://github.com/rhasspy/piper

After some brief research it seems the issue you're seeing may be a known bug in at least some versions/release of espeak-ng.

Here's some potentially related links if you'd like to dig deeper:

* "questions about mandarin data packet #1044": https://github.com/espeak-ng/espeak-ng/issues/1044

* "ESpeak NJ-1.51’s Mandarin pronunciation is corrupted #12952": https://github.com/nvaccess/nvda/issues/12952

* "The pronunciation of Mandarin Chinese using ESpeak NJ in NVDA is not normal #1028": https://github.com/espeak-ng/espeak-ng/issues/1028

* "When espeak-ng translates Chinese (cmn), IPA tone symbols are not output correctly #305": https://github.com/rhasspy/piper/issues/305

* "Please default ESpeak NG's voice role to 'Chinese (Mandarin, latin as Pinyin)' for Chinese to fix #12952 #13572": https://github.com/nvaccess/nvda/issues/13572

* "Cmn voice not correctly translated #1370": https://github.com/espeak-ng/espeak-ng/issues/1370

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