We have developed and released the family Ichigo-llama3s. This family is natively understanding audio and text input.
This model is a supervised fine-tuned (SFT) version of homebrewltd/Ichigo-llama3.1-s-base-v0.3, trained on over 1 billion tokens from the Instruction Speech WhisperVQ v4 dataset which built upon Instruction Speech WhisperVQ v3, adding multi-turn speech conversations and noise rejection capabilities for enhanced performance. As a result, the model demonstrates improved robustness against noisy environmental inputs and enhanced multi-turn conversation capabilities, making it more reliable in real-world applications.
Model developers Homebrew Research.
Input Text and sound.
Output Text.
Model Architecture Llama-3.
Language(s): English.
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.
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)
vq_model.ensure_whisper(device)
def audio_to_sound_tokens(audio_path, target_bandwidth=1.5, device=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|>'
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 Metrics Image: Below is a snapshot of the training loss curve visualized.

MMLU:
| Model | MMLU Score |
|---|---|
| llama3.1-instruct-8b | 69.40 |
| ichigo-llama3.1-s-v0.4 | 64.66 |
| ichigo-llama3.1-s-v0.3: phase 3 | 63.79 |
| ichigo-llama3.1-s-v0.3: phase 2 | 63.08 |
| ichigo-llama3.1-s-base-v0.3 | 42.11 |
| llama3.5-instruct-v0.2 | 50.27 |
AudioBench Eval:
| Model Bench | Open-hermes Instruction Audio (GPT-4-O judge 0:5) | Alpaca Instruction Audio (GPT-4-O judge 0:5) |
|---|---|---|
| Llama3.1-s-v2 | 3.45 | 3.53 |
| Ichigo-llama3.1-s v0.4 | 3.5 | 3.52 |
| Ichigo-llama3.1-s v0.3-phase2 -cp7000 | 3.42 | 3.62 |
| Ichigo-llama3.1-s v0.3-phase2-cplast | 3.31 | 3.6 |
| Ichigo-llama3.1-s v0.3-phase3 | 3.64 | 3.68 |
| Qwen2-audio-7B | 2.63 | 2.24 |
GPU Configuration: Cluster of 8x NVIDIA H100-SXM-80GB.
GPU Usage:
We utilize torchtune library for the latest FSDP2 training code implementation.
| Parameter | Instruction Fine-tuning |
|---|---|
| Epoch | 1 |
| Global batch size | 256 |
| Learning Rate | 7e-5 |
| Learning Scheduler | Cosine with warmup |
| Optimizer | Adam torch fused |
| Warmup Ratio | 0.01 |
| Weight Decay | 0.005 |
| Max Sequence Length | 4096 |
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}
We have developed and released the family Ichigo-llama3s. This family is natively understanding audio and text input.
This model is a supervised fine-tuned (SFT) version of homebrewltd/Ichigo-llama3.1-s-base-v0.3, trained on over 1 billion tokens from the Instruction Speech WhisperVQ v4 dataset which built upon Instruction Speech WhisperVQ v3, adding multi-turn speech conversations and noise rejection capabilities for enhanced performance. As a result, the model demonstrates improved robustness against noisy environmental inputs and enhanced multi-turn conversation capabilities, making it more reliable in real-world applications.
Model developers Homebrew Research.
Input Text and sound.
Output Text.
Model Architecture Llama-3.
Language(s): English.
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.
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)
vq_model.ensure_whisper(device)
def audio_to_sound_tokens(audio_path, target_bandwidth=1.5, device=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|>'
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 Metrics Image: Below is a snapshot of the training loss curve visualized.

MMLU:
| Model | MMLU Score |
|---|---|
| llama3.1-instruct-8b | 69.40 |
| ichigo-llama3.1-s-v0.4 | 64.66 |
| ichigo-llama3.1-s-v0.3: phase 3 | 63.79 |
| ichigo-llama3.1-s-v0.3: phase 2 | 63.08 |
| ichigo-llama3.1-s-base-v0.3 | 42.11 |
| llama3.5-instruct-v0.2 | 50.27 |
AudioBench Eval:
| Model Bench | Open-hermes Instruction Audio (GPT-4-O judge 0:5) | Alpaca Instruction Audio (GPT-4-O judge 0:5) |
|---|---|---|
| Llama3.1-s-v2 | 3.45 | 3.53 |
| Ichigo-llama3.1-s v0.4 | 3.5 | 3.52 |
| Ichigo-llama3.1-s v0.3-phase2 -cp7000 | 3.42 | 3.62 |
| Ichigo-llama3.1-s v0.3-phase2-cplast | 3.31 | 3.6 |
| Ichigo-llama3.1-s v0.3-phase3 | 3.64 | 3.68 |
| Qwen2-audio-7B | 2.63 | 2.24 |
GPU Configuration: Cluster of 8x NVIDIA H100-SXM-80GB.
GPU Usage:
We utilize torchtune library for the latest FSDP2 training code implementation.
| Parameter | Instruction Fine-tuning |
|---|---|
| Epoch | 1 |
| Global batch size | 256 |
| Learning Rate | 7e-5 |
| Learning Scheduler | Cosine with warmup |
| Optimizer | Adam torch fused |
| Warmup Ratio | 0.01 |
| Weight Decay | 0.005 |
| Max Sequence Length | 4096 |
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}