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cstr/MOSS-Audio-4B-Instruct-GGUF

sourceHugging Faceapache-2.0updated 2mo agoView on Hugging Face

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Model Card

MOSS-Audio-4B-Instruct -- GGUF (ggml-quantised)

GGUF / ggml conversions of `OpenMOSS-Team/MOSS-Audio-4B-Instruct` for use with crispasr --backend moss-audio from [CrispStrobe/CrispASR](https://github.com/CrispStrobe/CrispASR).

MOSS-Audio-4B-Instruct is OpenMOSS's ~4.6 B parameter audio-understanding model:

—First audio-understanding backend in CrispASR -- not just ASR but also audio QA, scene description, music analysis, meeting summarisation
—Mandarin + English speech recognition and audio understanding
—DeepStack cross-layer feature injection -- multi-resolution encoder taps at layers 8/16/24 injected into the LM's early layers for fine-grained prosody + semantic awareness
—Time-aware ASR with explicit time-marker tokens for word-level and sentence-level timestamps
—Apache-2.0 licence

Files

File	Size	Notes
`moss-audio-4b-instruct-f16.gguf`	9.73 GB	F16, full precision
`moss-audio-4b-instruct-q4_k.gguf`	2.75 GB	Q4_K -- recommended default

Quick Start

bash

# 1. Build the runtime
git clone https://github.com/CrispStrobe/CrispASR
cd CrispASR
cmake -G Ninja -B build -DCMAKE_BUILD_TYPE=Release -DGGML_CUDA=OFF
cmake --build build -j$(nproc) --target crispasr-cli

# 2. Download a quantisation
huggingface-cli download cstr/MOSS-Audio-4B-Instruct-GGUF \
    moss-audio-4b-instruct-q4_k.gguf --local-dir .

# 3. Transcribe audio
./build/bin/crispasr \
    -m moss-audio-4b-instruct-q4_k.gguf \
    -f your-audio.wav \
    --backend moss-audio -t 4

# 4. Audio understanding (custom prompt)
./build/bin/crispasr \
    -m moss-audio-4b-instruct-q4_k.gguf \
    -f your-audio.wav \
    --backend moss-audio \
    --prompt "Describe the sounds in this audio clip."

Verified end-to-end output

JFK sample (samples/jfk.wav, 11s):

And so, my fellow Americans, ask not what your country can do for you, ask what you can do for your country.

Verified on Q4K (3.8 GB, F16 encoder + Q4K LLM). All 6 crispasr-diff stages PASS at cos >= 0.999.

Architecture

Component	Details
Audio encoder	32-layer Whisper-style transformer (d=1280, 20 heads, head_dim=64, FFN=5120, GELU, LayerNorm, eps=1e-5)
Conv stem	3x Conv2d(stride=2, channels=480, kernel=3x3, pad=1) -> 8x temporal downsample (128 mel bins -> 16 freq bins)
Stem projection	Linear(480x16=7680 -> 1280) + sinusoidal positional embedding (max 1500 positions)
DeepStack taps	Encoder layers [8, 16, 24] -> 3 independent GatedMLP(1280 -> 8192 -> 2560, SiLU)
DeepStack injection	Residual add at LM layers [0, 1, 2] at audio-token positions
Audio adapter	GatedMLP(1280 -> 8192 -> 2560, SiLU) for final encoder output
LM backbone	36-layer Qwen3 (hidden=2560, 32 Q-heads / 8 KV-heads, head_dim=128, QK-norm, SwiGLU FFN=9728, RoPE theta=1M)
Output head	Linear(2560 -> 151936), untied from embedding
Vocab	151936 Qwen3 BPE (151643 regular + 293 special tokens)
Audio input	16 kHz mono, 128 mel bins, n_fft=400, hop=160
Audio tokens	12.5 Hz after 8x conv downsample, time markers every 2 seconds
Parameters	~4.6 B total (encoder ~650M + adapter/deepstack ~120M + LM ~3.8B)

Special tokens

Token	ID	Purpose
`<\	AUDIO\	>`	151654	Audio frame placeholder (replaced by encoder embeddings)
`<\	audio_bos\	>`	151669	Audio segment start marker
`<\	audio_eos\	>`	151670	Audio segment end marker
`<\	im_start\	>`	151644	Chat turn start
`<\	im_end\	>`	151645	Chat turn end / EOS

How this was made

1.Inspect the HF safetensors: 3 shards, 901 tensors total -- audio encoder (conv stem + 32 transformer layers + layernorm), audio adapter (1 GatedMLP), deepstack mergers (3 GatedMLPs), language model (embedding + 36 Qwen3 layers + final norm + lmhead).

1.Convert with `models/convert-moss-audio-to-gguf.py`: stream BF16 tensors one-at-a-time via safe_open, remap HF tensor names (audio_encoder.layers.N.self_attn.q_proj -> enc.blk.N.attn.q, deepstack_audio_merger_list.N.gate_proj -> deepstack.N.gate, language_model.layers.N.mlp.gate_proj -> llm.blk.N.ffn.gate, etc.), write F16 + F32 (norms/biases). BPE vocab + merges from vocab.json + merges.txt + added_tokens.json.

1.Quantize with crispasr-quantize: F16 -> Q4_K (2D+ tensors quantised, 1D biases/norms kept F32).

1.C++ runtime in `src/moss_audio.{h,cpp}`: GGUF mmap, encoder graph (conv stem + 32 WhisperEncoderLayers with bidirectional attention + DeepStack tap capture), adapter/merger GatedMLP graphs, per-layer DeepStack injection into LM via pre-scattered residuals, KV-cached Qwen3 decode with core_attn::kv_self_attn (QK-norm, RoPE, GQA), greedy decode with chat-template prompt builder.

Upstream

—Model: `OpenMOSS-Team/MOSS-Audio-4B-Instruct` (Apache-2.0)
—Code: `OpenMOSS/MOSS-Audio`
—Runtime: `CrispStrobe/CrispASR` branch feature/moss-audio

Provenance and EU AI Act Art. 53 note

—Upstream model: OpenMOSS-Team/MOSS-Audio-4B-Instruct — published by OpenMOSS-Team.
—Upstream licence: apache-2.0. This repository redistributes under the same terms; it grants no rights the upstream licence does not.
—What was done here: format conversion and/or quantisation only (GGUF/GGML). No training, no fine-tuning, no merging, no distillation, no change to architecture, vocabulary or capability. Only the numeric representation of the upstream weights differs.
—Training data: documented — where it is documented at all — by the upstream provider; see the upstream model card. No training data was used, added or selected by this repository. No training-content summary was found on the upstream model card at the time of writing; that documentation gap is upstream's and is not filled here.
—Provider status: under Regulation (EU) 2024/1689 the upstream authors remain the provider of this model. Converting the serialisation format does not make this repository the provider of a new general-purpose AI model, and no such claim is made. Questions about training content, copyright policy or model capability belong upstream.