README.md

---
license: cc-by-nc-4.0
language:
- de
base_model:
- HKUSTAudio/Llasa-1B-Multilingual
widget:
  - src: examples/no_speaker_example.wav

---
<img src="https://huggingface.co/MultiLlasa/Llasa-1B-Multilingual-German/resolve/main/cover.webp" alt="Llasa German Logo" width="800" style="margin-left:'auto' margin-right:'auto' display:'block'"/>


# Llasa-1B-Multilingual-German
<a target="_blank" href="https://huggingface.co/spaces/SebastianBodza/llasa-1b-tts-german">
  <img src="https://huggingface.co/datasets/huggingface/badges/raw/main/open-in-hf-spaces-sm.svg" alt="Open in HuggingFace"/>
</a>

<Gallery />


> This model was trained on top of [HKUSTAudio/Llasa-1B-Multilingual](https://huggingface.co/HKUSTAudio/Llasa-1B-Multilingual).

## Model Overview

This text-to-speech (TTS) model has been trained on a custom dataset representing **7,000 hours** of high-quality audio data. The audio data consisted of permissive podcasts, lectures and other OER data. 



## Training Details

- **Base Model:** HKUSTAudio/Llasa-1B-Multilingual
- **Dataset:** A custom dataset comprising **7,000 hours** of data.
- **Compute Resources:** The training was performed using **4x L40s GPUs**.
- **Raw Training Time:** Approximately **20 hours** not included the data preprocessing with xcodec2 (note: training was restarted after 3 crashes).
  
Huge thanks to Hugging Face for their generous GPU grant! 🤗


## 👨‍💻 Installation
First install the following pip packages: 
```bash
pip install xcodec2
pip install torch==2.6.0 torchaudio
```
Install it in the two steps given above! If you get the error message with "flex attention" make sure to install `torch==2.6.0 torchaudio`. If you get an torchaudio error, make sure to update and match it to the torch 2.6.0 version. 

## 🛠️ Usage
### 🎲 Random voice
A basic example using the Hugging Face Transformers:

```python
import os
from transformers import AutoTokenizer, AutoModelForCausalLM
import torch
import soundfile as sf

llasa_1b_german = 'MultiLlasa/Llasa-1B-Multilingual-German'  

# Loading the model 
tokenizer = AutoTokenizer.from_pretrained(llasa_1b_german)
model = AutoModelForCausalLM.from_pretrained(llasa_1b_german)
model.to('cuda')

# Load XCodec2 model
from xcodec2.modeling_xcodec2 import XCodec2Model
model_path = "HKUST-Audio/xcodec2"
Codec_model = XCodec2Model.from_pretrained(model_path)
Codec_model.cuda()

input_text = "\"Weißt du was, Hoppi\", sagte der weise Uhu, \"manchmal ist es gar nicht so wichtig, das Ende des Regenbogens zu finden. Das Schönste ist doch, dass wir alle zusammen dieses Abenteuer erleben!"


def extract_speech_ids(speech_tokens_str):
    speech_ids = []
    for token_str in speech_tokens_str:
        if token_str.startswith('<|s_') and token_str.endswith('|>'):
            num_str = token_str[4:-2]
            num = int(num_str)
            speech_ids.append(num)
        else:
            print(f"Unexpected token: {token_str}")
    return speech_ids

with torch.no_grad():
    formatted_text = f"<|TEXT_UNDERSTANDING_START|>{input_text}<|TEXT_UNDERSTANDING_END|>"
    
    chat = [
        {"role": "user", "content": "Convert the text to speech:" + formatted_text},
        {"role": "assistant", "content": "<|SPEECH_GENERATION_START|>"}
    ]

    input_ids = tokenizer.apply_chat_template(
        chat,
        tokenize=True,
        return_tensors='pt',
        continue_final_message=True
    )
    input_ids = input_ids.to('cuda')
    speech_end_id = tokenizer.convert_tokens_to_ids('<|SPEECH_GENERATION_END|>')

    outputs = model.generate(
        input_ids,
        max_length=2048,
        eos_token_id=speech_end_id,
        do_sample=True,
        top_p=1,
        temperature=0.8,
    )

    generated_ids = outputs[0][input_ids.shape[1]:-1]
    speech_tokens = tokenizer.batch_decode(generated_ids, skip_special_tokens=True)
    speech_tokens = extract_speech_ids(speech_tokens)
    speech_tokens = torch.tensor(speech_tokens).cuda().unsqueeze(0).unsqueeze(0)
    gen_wav = Codec_model.decode_code(speech_tokens)
    
    
sf.write("generation.wav", gen_wav[0, 0, :].cpu().numpy(), 16000)

```

### 🎯 Using a specific speaker

An example with speaker reference: 
```python 
import torch
import torchaudio
import tempfile
import soundfile as sf
from transformers import AutoTokenizer, AutoModelForCausalLM, pipeline

# Input your reference audio and optional the text
sample_audio_path = "male.wav"
sample_audio_text = None # Set it to none to use whisper for transcription
# Input the target text here
target_text = "Und apropos Spannungen und Unfälle, in Stuttgart gibt es auch einige Schlagzeilen. Die Polizei sucht Zeugen, nachdem in der Stadt mehrere Autoscheiben eingeschlagen wurden. Und gestern kam es im Stuttgarter Osten zu einer Verfolgungsjagd mit einer jungen BMW-Fahrerin, die vor einer Polizeistreife geflüchtet ist."
output_filename = "no_speaker_example.wav"


#### Do not edit below ####
llasa_model_name = "MultiLlasa/Llasa-1B-Multilingual-German"
tokenizer = AutoTokenizer.from_pretrained(llasa_model_name)
model = AutoModelForCausalLM.from_pretrained(llasa_model_name)
model.to("cuda")

from xcodec2.modeling_xcodec2 import XCodec2Model
codec_model_path = "HKUST-Audio/xcodec2"
Codec_model = XCodec2Model.from_pretrained(codec_model_path)
Codec_model.cuda()

whisper_turbo_pipe = pipeline(
    "automatic-speech-recognition",
    model="openai/whisper-large-v3-turbo",
    torch_dtype=torch.float16,
    device="cuda",
)

def ids_to_speech_tokens(speech_ids):
    speech_tokens_str = []
    for speech_id in speech_ids:
        speech_tokens_str.append(f"<|s_{speech_id}|>")
    return speech_tokens_str

waveform, sample_rate = torchaudio.load(sample_audio_path)

max_secs = 15
if len(waveform[0]) / sample_rate > 15:
    print("Warning: Trimming audio to first 15secs.")
    waveform = waveform[:, : sample_rate * 15]
    waveform = torch.nn.functional.pad( waveform, (0, int(sample_rate * 0.5)), "constant", 0)

if waveform.size(0) > 1:
    waveform = torch.mean(waveform, dim=0, keepdim=True)

prompt_wav = torchaudio.transforms.Resample(orig_freq=sample_rate, new_freq=16000)(waveform)

if sample_audio_text is None:
    print("Transcribing audio...")
    transcription = whisper_turbo_pipe(waveform[0].numpy())["text"].strip()
else:
    transcription = sample_audio_text

print("Transcription:", transcription)

if len(target_text) == 0:
    raise ValueError("Target text must be provided!")
elif len(target_text) > 500:
    print("Text is too long; trimming to first 500 characters.")
    target_text = target_text[:500]

input_text = transcription + " " + target_text

with torch.no_grad():
    vq_code_prompt = Codec_model.encode_code(input_waveform=prompt_wav)
    vq_code_prompt = vq_code_prompt[0, 0, :]
    speech_ids_prefix = ids_to_speech_tokens(vq_code_prompt)

    formatted_text = f"<|TEXT_UNDERSTANDING_START|>{input_text}<|TEXT_UNDERSTANDING_END|>"

    chat = [
        {"role": "user", "content": "Convert the text to speech:" + formatted_text},
        {"role": "assistant", "content": "<|SPEECH_GENERATION_START|>" + "".join(speech_ids_prefix)}
        ]

    input_ids = tokenizer.apply_chat_template(chat, tokenize=True, return_tensors="pt", continue_final_message=True)
    input_ids = input_ids.to("cuda")
    speech_end_id = tokenizer.convert_tokens_to_ids("<|SPEECH_GENERATION_END|>")

    outputs = model.generate(
        input_ids,
        max_length=2048, 
        eos_token_id=speech_end_id,
        do_sample=True,
        top_p=1,
        temperature=0.8,
        min_new_tokens=4, # Fix so the model does not directly stop 
    )

    generated_ids = outputs[0][input_ids.shape[1] - len(speech_ids_prefix) : -1]

    speech_tokens = tokenizer.batch_decode(generated_ids, skip_special_tokens=True)
    speech_tokens = extract_speech_ids(speech_tokens)
    speech_tokens = torch.tensor(speech_tokens).cuda().unsqueeze(0).unsqueeze(0)

    gen_wav = Codec_model.decode_code(speech_tokens)
    gen_wav = gen_wav[:, :, prompt_wav.shape[1] :]
    sf.write(output_filename, gen_wav[0, 0, :].cpu().numpy(), 16000)
```


## Tips
- With a reference speaker, audio glitches can happen. Try to increase the temperature to get better results. 

## License

This project is licensed under the [CC-BY-NC-4.0 license](https://creativecommons.org/licenses/by-nc/4.0/).

## Acknowledgments

- **Hugging Face:** Thanks for the grant that made this project possible.
* [**HKUSTAudio:**](https://huggingface.co/HKUSTAudio/Llasa-1B-Multilingual) for providing the model open source and a great inference, training and preprocessing (xcodec2) script!