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--- |
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datasets: |
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- homebrewltd/instruction-speech-whispervq-v2 |
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language: |
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- en |
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license: apache-2.0 |
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tags: |
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- sound language model |
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--- |
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# QuantFactory/llama3.1-s-instruct-v0.2-GGUF |
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This is quantized version of [homebrewltd/llama3.1-s-instruct-v0.2](https://huggingface.co/homebrewltd/llama3.1-s-instruct-v0.2) created using llama.cpp |
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# Original Model Card |
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## Model Details |
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We have developed and released the family [llama3s](https://huggingface.co/collections/homebrew-research/llama3-s-669df2139f0576abc6eb7405). This family is natively understanding audio and text input. |
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We expand the Semantic tokens experiment with WhisperVQ as a tokenizer for audio files from [homebrewltd/llama3.1-s-base-v0.2](https://huggingface.co/homebrewltd/llama3.1-s-base-v0.2) with nearly 1B tokens from [Instruction Speech WhisperVQ v2](https://huggingface.co/datasets/homebrewltd/instruction-speech-whispervq-v2) dataset. |
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**Model developers** Homebrew Research. |
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**Input** Text and sound. |
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**Output** Text. |
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**Model Architecture** Llama-3. |
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**Language(s):** English. |
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## Intended Use |
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**Intended Use Cases** This family is primarily intended for research applications. This version aims to further improve the LLM on sound understanding capabilities. |
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**Out-of-scope** The use of llama3-s in any manner that violates applicable laws or regulations is strictly prohibited. |
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## How to Get Started with the Model |
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Try this model using [Google Colab Notebook](https://colab.research.google.com/drive/18IiwN0AzBZaox5o0iidXqWD1xKq11XbZ?usp=sharing). |
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First, we need to convert the audio file to sound tokens |
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```python |
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device = "cuda" if torch.cuda.is_available() else "cpu" |
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if not os.path.exists("whisper-vq-stoks-medium-en+pl-fixed.model"): |
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hf_hub_download( |
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repo_id="jan-hq/WhisperVQ", |
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filename="whisper-vq-stoks-medium-en+pl-fixed.model", |
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local_dir=".", |
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) |
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vq_model = RQBottleneckTransformer.load_model( |
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"whisper-vq-stoks-medium-en+pl-fixed.model" |
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).to(device) |
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def audio_to_sound_tokens(audio_path, target_bandwidth=1.5, device=device): |
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vq_model.ensure_whisper(device) |
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wav, sr = torchaudio.load(audio_path) |
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if sr != 16000: |
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wav = torchaudio.functional.resample(wav, sr, 16000) |
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with torch.no_grad(): |
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codes = vq_model.encode_audio(wav.to(device)) |
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codes = codes[0].cpu().tolist() |
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result = ''.join(f'<|sound_{num:04d}|>' for num in codes) |
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return f'<|sound_start|>{result}<|sound_end|>' |
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def audio_to_sound_tokens_transcript(audio_path, target_bandwidth=1.5, device=device): |
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vq_model.ensure_whisper(device) |
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wav, sr = torchaudio.load(audio_path) |
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if sr != 16000: |
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wav = torchaudio.functional.resample(wav, sr, 16000) |
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with torch.no_grad(): |
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codes = vq_model.encode_audio(wav.to(device)) |
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codes = codes[0].cpu().tolist() |
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result = ''.join(f'<|sound_{num:04d}|>' for num in codes) |
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return f'<|reserved_special_token_69|><|sound_start|>{result}<|sound_end|>' |
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``` |
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Then, we can inference the model the same as any other LLM. |
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```python |
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def setup_pipeline(model_path, use_4bit=False, use_8bit=False): |
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tokenizer = AutoTokenizer.from_pretrained(model_path) |
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model_kwargs = {"device_map": "auto"} |
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if use_4bit: |
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model_kwargs["quantization_config"] = BitsAndBytesConfig( |
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load_in_4bit=True, |
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bnb_4bit_compute_dtype=torch.bfloat16, |
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bnb_4bit_use_double_quant=True, |
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bnb_4bit_quant_type="nf4", |
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) |
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elif use_8bit: |
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model_kwargs["quantization_config"] = BitsAndBytesConfig( |
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load_in_8bit=True, |
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bnb_8bit_compute_dtype=torch.bfloat16, |
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bnb_8bit_use_double_quant=True, |
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) |
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else: |
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model_kwargs["torch_dtype"] = torch.bfloat16 |
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model = AutoModelForCausalLM.from_pretrained(model_path, **model_kwargs) |
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return pipeline("text-generation", model=model, tokenizer=tokenizer) |
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def generate_text(pipe, messages, max_new_tokens=64, temperature=0.0, do_sample=False): |
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generation_args = { |
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"max_new_tokens": max_new_tokens, |
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"return_full_text": False, |
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"temperature": temperature, |
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"do_sample": do_sample, |
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} |
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output = pipe(messages, **generation_args) |
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return output[0]['generated_text'] |
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# Usage |
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llm_path = "homebrewltd/llama3.1-s-instruct-v0.2" |
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pipe = setup_pipeline(llm_path, use_8bit=True) |
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``` |
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## Training process |
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**Training Metrics Image**: Below is a snapshot of the training loss curve visualized. |
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### Hardware |
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**GPU Configuration**: Cluster of 8x NVIDIA H100-SXM-80GB. |
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**GPU Usage**: |
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- **Continual Training**: 6 hours. |
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### Training Arguments |
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We utilize [torchtune](https://github.com/pytorch/torchtune) library for the latest FSDP2 training code implementation. |
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| Parameter | Continual Training | |
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|----------------------------|-------------------------| |
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| **Epoch** | 1 | |
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| **Global batch size** | 128 | |
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| **Learning Rate** | 0.5e-4 | |
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| **Learning Scheduler** | Cosine with warmup | |
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| **Optimizer** | Adam torch fused | |
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| **Warmup Ratio** | 0.01 | |
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| **Weight Decay** | 0.005 | |
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| **Max Sequence Length** | 512 | |
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## Examples |
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1. Good example: |
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<details> |
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<summary>Click to toggle Example 1</summary> |
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``` |
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``` |
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</details> |
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<details> |
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<summary>Click to toggle Example 2</summary> |
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``` |
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``` |
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</details> |
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2. Misunderstanding example: |
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<details> |
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<summary>Click to toggle Example 3</summary> |
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``` |
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``` |
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</details> |
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3. Off-tracked example: |
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<details> |
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<summary>Click to toggle Example 4</summary> |
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``` |
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``` |
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</details> |
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## Citation Information |
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**BibTeX:** |
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``` |
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@article{Llama3-S: Sound Instruction Language Model 2024, |
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title={Llama3-S}, |
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author={Homebrew Research}, |
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year=2024, |
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month=August}, |
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url={https://huggingface.co/homebrewltd/llama3.1-s-2024-08-20} |
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``` |
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## Acknowledgement |
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- **[WhisperSpeech](https://github.com/collabora/WhisperSpeech)** |
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- **[Meta-Llama-3.1-8B-Instruct ](https://huggingface.co/meta-llama/Meta-Llama-3.1-8B-Instruct)** |
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