MOSS-Transcribe-Diarize

MOSS-Transcribe-Diarize 0.9B is an end-to-end audio understanding model for long-form multi-speaker transcription, diarization, timestamps, and acoustic event awareness.

Given an audio or video file, the model generates a compact speaker-aware transcript in one pass, including timestamps and anonymous speaker labels such as [S01], [S02], and beyond.

News

  • 2026-07-09: Released MOSS-Transcribe-Diarize 0.9B.

Contents

Introduction

MOSS-Transcribe-Diarize 0.9B turns real-world long-form audio into structured, speaker-aware transcripts in one pass. Instead of stitching together separate ASR and diarization systems, it jointly performs speech transcription and speaker diarization, producing time-aligned text with consistent speaker labels.

The model is built for meetings, calls, podcasts, interviews, lectures, videos, and other long or messy multi-speaker recordings. It can also emit acoustic event annotations, giving downstream systems a richer view of what happened, who spoke, and when.

Core capabilities:

  • Long-form transcription: Converts long audio or video recordings into timestamped text.
  • Speaker-aware diarization: Assigns anonymous speaker labels such as [S01] and [S02] without a separate diarization pipeline.
  • Promptable generation: Supports custom transcription instructions, hotwords, and acoustic event annotations.

Model Architecture

MOSS-Transcribe-Diarize 0.9B model architecture

Component Specification
Text backbone Qwen3-0.6B style causal decoder
Audio encoder Whisper-Medium encoder configuration
Audio frontend WhisperFeatureExtractor, 16 kHz, 80 mel bins, 30 s chunks
Audio-text bridge 4x temporal merge + MLP adaptor
Fusion Audio features replace <|audio_pad|> embeddings via masked_scatter
Output format Compact [start][Sxx]text[end] transcript with speaker tags such as [S01]

This Hugging Face repository includes the custom Transformers remote code required to load the model with trust_remote_code=True.

Evaluation

We evaluate MOSS-Transcribe-Diarize using three objective metrics: Character Error Rate (CER), concatenated minimum-permutation Character Error Rate (cpCER), and Delta-cp. Lower is better for all metrics. A dash (-) indicates that the result is unavailable.

Model AISHELLโ€‘4 Alimeeting Podcast Movies
CERโ†“cpCERโ†“ฮ”cpโ†“ CERโ†“cpCERโ†“ฮ”cpโ†“ CERโ†“cpCERโ†“ฮ”cpโ†“ CERโ†“cpCERโ†“ฮ”cpโ†“
Doubao 18.1827.869.68 25.2537.5712.31 7.9310.542.61 9.9430.8820.94
ElevenLabs 19.5837.9518.36 25.7036.6910.99 8.5011.342.85 11.4917.856.37
GPT-4o --- --- --- 14.3723.679.31
Gemini 2.5 Pro 42.7053.4210.72 27.4341.6414.21 7.3810.232.85 15.4624.158.69
Gemini 3 Pro 22.7527.434.68 26.7532.846.09 --- 8.6214.736.11
VIBEVOICE ASR 21.4024.993.59 27.4029.331.93 27.9448.3020.36 14.5942.5427.94
MOSS Transcribe Diarize 0.9B 14.8415.830.99 24.8622.17-2.69 5.977.371.40 6.3612.766.40
MOSS Transcribe Diarize Pro 13.7814.020.24 18.2213.94-4.27 4.466.972.51 5.8611.785.92

Quickstart

Environment Setup

Use a clean Python environment. The model uses custom Transformers code, so load the model and processor with trust_remote_code=True.

conda create -n moss-transcribe-diarize python=3.12 -y
conda activate moss-transcribe-diarize

git clone https://github.com/OpenMOSS/MOSS-Transcribe-Diarize.git
cd MOSS-Transcribe-Diarize

pip install --index-url https://download.pytorch.org/whl/cu128 torch torchaudio
pip install -e .

The GitHub package provides helper utilities such as audio/video loading, transcription message construction, transcript parsing, CLI inference, and the subtitle web app. The model weights and remote-code model files are loaded from this Hugging Face repository.

Python Usage

import torch
from transformers import AutoModelForCausalLM, AutoProcessor

from moss_transcribe_diarize import parse_transcript
from moss_transcribe_diarize.inference_utils import (
    build_transcription_messages,
    generate_transcription,
    resolve_device,
)

model_id = "OpenMOSS-Team/MOSS-Transcribe-Diarize"
audio_path = "audio.wav"

device = resolve_device("auto")
dtype = torch.bfloat16 if device.type == "cuda" else torch.float32

model = AutoModelForCausalLM.from_pretrained(
    model_id,
    trust_remote_code=True,
    dtype="auto",
).to(dtype=dtype).to(device).eval()

processor = AutoProcessor.from_pretrained(
    model_id,
    trust_remote_code=True,
)

messages = build_transcription_messages(audio_path)
result = generate_transcription(
    model,
    processor,
    messages,
    max_new_tokens=2048,
    do_sample=False,
    device=device,
    dtype=dtype,
)

print(result["text"])

for segment in parse_transcript(result["text"]):
    print(segment.start, segment.end, segment.speaker, segment.text)

The message flow follows the common Qwen multimodal pattern:

  1. processor.apply_chat_template(messages, tokenize=False) renders text with audio placeholders.
  2. The helper utilities load audio waveforms from the same messages.
  3. processor(text=text, audio=audios) computes Whisper input features and expands audio placeholders.
  4. model.generate(...) produces timestamped transcription and diarization text.

Custom Prompt and Hotwords

The default prompt is optimized for timestamped transcription and speaker diarization:

่ฏทๅฐ†้Ÿณ้ข‘่ฝฌๅ†™ไธบๆ–‡ๆœฌ๏ผŒๆฏไธ€ๆฎต้œ€ไปฅ่ตทๅง‹ๆ—ถ้—ดๆˆณๅ’Œ่ฏด่ฏไบบ็ผ–ๅท๏ผˆ[S01]ใ€[S02]ใ€[S03]โ€ฆ๏ผ‰ๅผ€ๅคด๏ผŒๆญฃๆ–‡ไธบๅฏนๅบ”็š„่ฏญ้Ÿณๅ†…ๅฎน๏ผŒๅนถๅœจๆฎตๆœซๆ ‡ๆณจ็ป“ๆŸๆ—ถ้—ดๆˆณ๏ผŒไปฅๆธ…ๆ™ฐๆ ‡ๆ˜Ž่ฏฅๆฎต่ฏญ้Ÿณ่Œƒๅ›ดใ€‚

To add hotwords, append a short hint to the default prompt:

่ฏทๅฐ†้Ÿณ้ข‘่ฝฌๅ†™ไธบๆ–‡ๆœฌ๏ผŒๆฏไธ€ๆฎต้œ€ไปฅ่ตทๅง‹ๆ—ถ้—ดๆˆณๅ’Œ่ฏด่ฏไบบ็ผ–ๅท๏ผˆ[S01]ใ€[S02]ใ€[S03]โ€ฆ๏ผ‰ๅผ€ๅคด๏ผŒๆญฃๆ–‡ไธบๅฏนๅบ”็š„่ฏญ้Ÿณๅ†…ๅฎน๏ผŒๅนถๅœจๆฎตๆœซๆ ‡ๆณจ็ป“ๆŸๆ—ถ้—ดๆˆณ๏ผŒไปฅๆธ…ๆ™ฐๆ ‡ๆ˜Ž่ฏฅๆฎต่ฏญ้Ÿณ่Œƒๅ›ดใ€‚็ƒญ่ฏๆ็คบ๏ผš็ƒญ่ฏ1, ็ƒญ่ฏ2, ็ƒญ่ฏ3

More prompt recipes are available in the GitHub repository: https://github.com/OpenMOSS/MOSS-Transcribe-Diarize/blob/main/examples/prompts.md

Serve with vLLM and SGLang

MOSS-Transcribe-Diarize supports vLLM serving through the OpenAI-compatible transcription API. Use a pinned vLLM nightly build that includes the MOSS-Transcribe-Diarize model registration. Choose one of the following commands: for CUDA 12 environments, use cu129; for CUDA 13 environments, use cu130.

uv pip install -U vllm \
  --torch-backend=auto \
  --extra-index-url https://wheels.vllm.ai/68b4a1d582818e67adc903bf1b8fc5a5447da2fa/cu129

or:

uv pip install -U vllm \
  --torch-backend=auto \
  --extra-index-url https://wheels.vllm.ai/68b4a1d582818e67adc903bf1b8fc5a5447da2fa/cu130
vllm serve OpenMOSS-Team/MOSS-Transcribe-Diarize --trust-remote-code
curl http://localhost:8000/v1/audio/transcriptions \
  -F model="OpenMOSS-Team/MOSS-Transcribe-Diarize" \
  -F file=@"audio.wav" \
  -F response_format="json" \
  -F temperature="0"

The recommended way to serve MOSS-Transcribe-Diarize is SGLang Omni through the OpenAI-compatible /v1/audio/transcriptions endpoint. Install sglang-omni by following the installation guide, then download the model:

hf download OpenMOSS-Team/MOSS-Transcribe-Diarize

Serve the model:

sgl-omni serve \
  --model-path OpenMOSS-Team/MOSS-Transcribe-Diarize \
  --port 8000 \
  --max-running-requests 16 \
  --cuda-graph-max-bs 16 \
  --mem-fraction-static 0.80

Use response_format=verbose_json when you need parsed speaker segments. json returns the raw transcript text only.

curl -X POST http://localhost:8000/v1/audio/transcriptions \
  -F model=OpenMOSS-Team/MOSS-Transcribe-Diarize \
  -F file=@audio.wav \
  -F response_format=verbose_json
import requests

with open("audio.wav", "rb") as f:
    resp = requests.post(
        "http://localhost:8000/v1/audio/transcriptions",
        data={
            "model": "OpenMOSS-Team/MOSS-Transcribe-Diarize",
            "response_format": "verbose_json",
        },
        files={"file": ("audio.wav", f, "audio/wav")},
        timeout=300,
    )

resp.raise_for_status()
payload = resp.json()
print(payload["text"])
for segment in payload.get("segments", []):
    print(f"[{segment['start']:.2f}-{segment['end']:.2f}] {segment['text']}")

For longer multi-speaker audio, raise max_new_tokens so the decoder can finish the full diarized transcript:

curl -X POST http://localhost:8000/v1/audio/transcriptions \
  -F model=OpenMOSS-Team/MOSS-Transcribe-Diarize \
  -F file=@audio.wav \
  -F response_format=verbose_json \
  -F max_new_tokens=65536
Parameter Type Default Description
file file required Audio file uploaded as multipart form data
model string server default Model identifier
language string unset Optional language hint
response_format string json json, verbose_json, or text
temperature float model default (0.0) Sampling temperature
max_new_tokens int 5120 Max generated tokens; raise for long audio, for example 65536
prompt string unset Optional instruction override; omit to use the built-in transcribe+diarize prompt

For benchmarking, performance numbers, and implementation details, see the SGLang Omni cookbook. The following single-H100 results are reported for short- and long-sequence multi-speaker ASR tasks.

movies short-sequence ASR:

Concurrency Throughput (req/s) Mean latency (s) RTF mean audio_s/s
1 2.57 0.388 0.0612 29.76
2 4.89 0.409 0.0659 56.55
4 6.62 0.513 0.0790 76.64
8 6.80 0.533 0.0810 78.70
16 7.08 0.659 0.0922 81.98

aishell4_long long-sequence ASR:

Concurrency Throughput (req/s) Mean latency (s) RTF mean audio_s/s
1 0.022 45.2 0.0197 50.64
2 0.032 60.7 0.0265 74.25
4 0.036 105.6 0.0461 81.64
8 0.040 172.6 0.0754 90.62
16 0.043 282.8 0.1237 98.83

Subtitle Web App

The source package includes a local subtitle workflow for upload, review, subtitle export, and optional FFmpeg burn-in:

mtd-subtitle-web \
  --model OpenMOSS-Team/MOSS-Transcribe-Diarize \
  --host 127.0.0.1 \
  --port 7860

Open http://127.0.0.1:7860, upload an audio/video file, review the parsed subtitle segments, then download JSON/SRT/ASS or burn an MP4 if ffmpeg and ffprobe are available on PATH.

For batch processing:

mtd-subtitle /path/to/input.mp4 \
  --model OpenMOSS-Team/MOSS-Transcribe-Diarize \
  --out-dir runs/example \
  --render

Output Format

The canonical output format is:

[start_time][Sxx]transcribed speech[end_time]

Example:

[0.48][S01]Welcome everyone[1.66][12.26][S02]The new transcription pipeline is ready for evaluation[13.81][14.36][S01]Great, include the diarization results in the report[18.76]

In this format:

  • start_time and end_time are timestamps in seconds.
  • [S01], [S02], and similar labels are anonymous model-generated speaker labels.
  • Speaker labels are relative labels within the input audio and should not be interpreted as real speaker identities.

More Information

License

MOSS-Transcribe-Diarize 0.9B is licensed under the Apache License 2.0.

Citation

If you use MOSS-Transcribe-Diarize 0.9B, please cite the technical report:

@misc{moss_transcribe_diarize_2026,
  title={MOSS Transcribe Diarize Technical Report},
  author={{MOSI.AI}},
  year={2026},
  eprint={2601.01554},
  archivePrefix={arXiv},
  primaryClass={cs.SD},
  url={https://arxiv.org/abs/2601.01554}
}
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