GGUF
English
sound language model
Inference Endpoints
conversational
aashish1904's picture
Upload README.md with huggingface_hub
c2b956e verified
|
raw
history blame
6.51 kB
metadata
datasets:
  - homebrewltd/instruction-speech-whispervq-v2
language:
  - en
license: apache-2.0
tags:
  - sound language model

QuantFactory/llama3.1-s-instruct-v0.2-GGUF

This is quantized version of homebrewltd/llama3.1-s-instruct-v0.2 created using llama.cpp

Original Model Card

Model Details

We have developed and released the family llama3s. This family is natively understanding audio and text input.

We expand the Semantic tokens experiment with WhisperVQ as a tokenizer for audio files from homebrewltd/llama3.1-s-base-v0.2 with nearly 1B tokens from Instruction Speech WhisperVQ v2 dataset.

Model developers Homebrew Research.

Input Text and sound.

Output Text.

Model Architecture Llama-3.

Language(s): English.

Intended Use

Intended Use Cases This family is primarily intended for research applications. This version aims to further improve the LLM on sound understanding capabilities.

Out-of-scope The use of llama3-s in any manner that violates applicable laws or regulations is strictly prohibited.

How to Get Started with the Model

Try this model using Google Colab Notebook.

First, we need to convert the audio file to sound tokens

device = "cuda" if torch.cuda.is_available() else "cpu"
if not os.path.exists("whisper-vq-stoks-medium-en+pl-fixed.model"):
    hf_hub_download(
        repo_id="jan-hq/WhisperVQ",
        filename="whisper-vq-stoks-medium-en+pl-fixed.model",
        local_dir=".",
    )
vq_model = RQBottleneckTransformer.load_model(
        "whisper-vq-stoks-medium-en+pl-fixed.model"
    ).to(device)
def audio_to_sound_tokens(audio_path, target_bandwidth=1.5, device=device):
    vq_model.ensure_whisper(device)

    wav, sr = torchaudio.load(audio_path)
    if sr != 16000:
        wav = torchaudio.functional.resample(wav, sr, 16000)
    with torch.no_grad():
        codes = vq_model.encode_audio(wav.to(device))
        codes = codes[0].cpu().tolist()

    result = ''.join(f'<|sound_{num:04d}|>' for num in codes)
    return f'<|sound_start|>{result}<|sound_end|>'

def audio_to_sound_tokens_transcript(audio_path, target_bandwidth=1.5, device=device):
    vq_model.ensure_whisper(device)

    wav, sr = torchaudio.load(audio_path)
    if sr != 16000:
        wav = torchaudio.functional.resample(wav, sr, 16000)
    with torch.no_grad():
        codes = vq_model.encode_audio(wav.to(device))
        codes = codes[0].cpu().tolist()

    result = ''.join(f'<|sound_{num:04d}|>' for num in codes)
    return f'<|reserved_special_token_69|><|sound_start|>{result}<|sound_end|>'

Then, we can inference the model the same as any other LLM.

def setup_pipeline(model_path, use_4bit=False, use_8bit=False):
    tokenizer = AutoTokenizer.from_pretrained(model_path)

    model_kwargs = {"device_map": "auto"}

    if use_4bit:
        model_kwargs["quantization_config"] = BitsAndBytesConfig(
            load_in_4bit=True,
            bnb_4bit_compute_dtype=torch.bfloat16,
            bnb_4bit_use_double_quant=True,
            bnb_4bit_quant_type="nf4",
        )
    elif use_8bit:
        model_kwargs["quantization_config"] = BitsAndBytesConfig(
            load_in_8bit=True,
            bnb_8bit_compute_dtype=torch.bfloat16,
            bnb_8bit_use_double_quant=True,
        )
    else:
        model_kwargs["torch_dtype"] = torch.bfloat16

    model = AutoModelForCausalLM.from_pretrained(model_path, **model_kwargs)

    return pipeline("text-generation", model=model, tokenizer=tokenizer)

def generate_text(pipe, messages, max_new_tokens=64, temperature=0.0, do_sample=False):
    generation_args = {
        "max_new_tokens": max_new_tokens,
        "return_full_text": False,
        "temperature": temperature,
        "do_sample": do_sample,
    }

    output = pipe(messages, **generation_args)
    return output[0]['generated_text']

# Usage
llm_path = "homebrewltd/llama3.1-s-instruct-v0.2"
pipe = setup_pipeline(llm_path, use_8bit=True)

Training process

Training Metrics Image: Below is a snapshot of the training loss curve visualized.

training_

Hardware

GPU Configuration: Cluster of 8x NVIDIA H100-SXM-80GB. GPU Usage:

  • Continual Training: 6 hours.

Training Arguments

We utilize torchtune library for the latest FSDP2 training code implementation.

Parameter Continual Training
Epoch 1
Global batch size 128
Learning Rate 0.5e-4
Learning Scheduler Cosine with warmup
Optimizer Adam torch fused
Warmup Ratio 0.01
Weight Decay 0.005
Max Sequence Length 512

Examples

  1. Good example:
Click to toggle Example 1

Click to toggle Example 2

  1. Misunderstanding example:
Click to toggle Example 3

  1. Off-tracked example:
Click to toggle Example 4

Citation Information

BibTeX:

@article{Llama3-S: Sound Instruction Language Model 2024,
  title={Llama3-S},
  author={Homebrew Research},
  year=2024,
  month=August},
  url={https://huggingface.co/homebrewltd/llama3.1-s-2024-08-20}

Acknowledgement