Model reference · open weights
moss-tts-local-transformer-voice-acting-sft3 is an open-weight audio or speech model from laion. moss-tts-local-transformer-4.55b-voice-acting-v2-sft3 (BF16) weighs 8.3 GB; the smallest configuration that runs it is RTX 3060 12 GB.
For self-hosting, compare this build's requirements below with the GPU you plan to rent.
Summary of the laion/moss-tts-local-transformer-4.55b-voice-acting-v2-sft3 model card, 2026-10-01
What it is
| Released by | laion |
|---|---|
| Released | 2026-08-26 |
| Parameters | 4.1B |
| VRAM | 8.3 GB for the weights |
What it runs on
| Card | One stream | Memory |
|---|---|---|
| RTX 3060 12 GB | fits | 11.6 GB |
| RTX 4060 Ti 16 GB | fits | 15.4 GB |
| RTX 3090 24 GB | fits | 23.4 GB |
| RTX 4090 24 GB | fits | 23.4 GB |
| RTX 5090 32 GB | fits | 31.0 GB |
| L40S 48 GB | fits | 44.0 GB |
| A100 80 GB | fits | 78.2 GB |
| H100 80 GB | fits | 78.1 GB |
| RTX PRO 6000 Blackwell 96 GB | fits | 93.8 GB |
| DGX Spark (GB10) 128 GB unified | fits | 107 GB |
| H200 141 GB | fits | 138 GB |
| B200 180 GB | fits | 176 GB |
From the model card
A supervised fine-tune of
laion/moss-tts-local-transformer-4.55b-voice-acting-v2
that adds explicit control over timing — per-sentence durations, pauses, and vocal bursts with
their lengths — and inline delivery directions that say how to perform each line.
4.13 B trainable parameters: a ~4 B semantic transformer (36 layers), a ~550 M local "talker" transformer, and 12 audio LM heads over a 12-codebook audio tokenizer at 12.5 frames per second (one frame = 80 ms).
Trained on 398,282 rows selected as the strongest examples of each of 40 emotions and each VoiceNet dimension, 2 epochs, 712 steps on 32 nodes. Evaluated by generating 320 clips and scoring them, not by validation loss — this project has repeatedly seen training metrics point the wrong way.
| round 2 | round 3 | |
|---|---|---|
| word error rate on direction-carrying prompts | 0.447 | 0.099 |
| duration error, median | 0.100 s | 0.080 s |
| clips within 0.5 s of the requested length | 92.8 % | 100 % |
| vocal-burst hit rate | 0.516 | 0.666 |
Round 2 had accidentally dropped the delivery directions its predecessor was trained with, and a model that has never seen a direction falls apart when given one — word error rate 0.48–0.51 on such prompts, for every round-2 model. Round 3 trained them back in. Timing control is solved.
Emotional intensity is not. Asked for percentile 0.90–0.98 of a named emotion, this model reaches about 0.35. Several objectives were tried against that — GRPO with a group-relative reward, DPO with contrastive pairs, DPO with symmetric instruction-conditioned pairs — and none moved it. The emotion adapters below are the only thing that has, and even they are not selective enough to merge in blindly. This is documented honestly in the technical report.
import torch, torchaudio
from transformers import AutoProcessor, AutoModel
BASE = "laion/moss-tts-local-transformer-4.55b-voice-acting-v2-sft3"
proc = AutoProcessor.from_pretrained(BASE, trust_remote_code=True)
model = AutoModel.from_pretrained(BASE, trust_remote_code=True,
dtype=torch.bfloat16, attn_implementation="sdpa").cuda().eval()
prompt = open("prompt.txt").read() # the block, see "How to prompt" below
um = {"role": "user", "content": prompt, "audio_codes_list": []}
b = proc([[um]], mode="generation")
with torch.no_grad():
out = model.generate(input_ids=b["input_ids"].cuda(),
attention_mask=b["attention_mask"].cuda(),
max_new_frames=340, do_sample=True,
audio_temperature=1.0, audio_top_p=0.95, audio_top_k=50,
audio_repetition_penalty=1.0)
# codes -> waveform. Use the processor's own decoder: calling the audio tokenizer directly, or
# reshaping its output, yields a two-channel result that flattens into audio at HALF SPEED and
# still sounds like speech. This project lost a whole corpus to that once.
wav = proc.decode_audio_codes([out_codes], return_stereo=False)[0].reshape(-1).float().cpu()
torchaudio.save("out.flac", wav[None], int(proc.model_config.sampling_rate), format="flac")
from peft import PeftModel
# one adapter
model = PeftModel.from_pretrained(model, "laion/moss-va-sft3-dpo-lora")
# several, each with its own weight -- the usual case: identity from a voice adapter,
# affect from an emotion adapter, general quality from the DPO adapter.
#
# NOTE: `add_weighted_adapter(..., combination_type="linear")` does NOT work here. It raises
# `ValueError: All adapters must have the same r value`, because the DPO adapter is rank 64 and
# the voice / emotion adapters are rank 16. Activate them together instead and scale each one.
model = PeftModel.from_pretrained(model, "", adapter_name="dpo")
model.load_adapter("", adapter_name="voice")
model.load_adapter("", adapter_name="emo")
names = ["dpo", "voice", "emo"]
weights = {"dpo": 1.0, "voice": 1.0, "emo": 1.5} # 1.5 for emotion is the measured optimum
model.base_model.set_adapter(names) # the TUNER takes a list; PeftModel does not
model.active_adapter = names[0] # must stay a str or generate() indexes a list
for mod in model.modules():
sc = getattr(mod, "scaling", None)
if isinstance(sc, dict):
if not hasattr(mod, "_base_scaling"):
mod._base_scaling = dict(sc)
for k in sc:
if k in weights:
sc[k] = mod._base_scaling[k] * weights[k]
A LoRA layer computes h + scaling · B(A(x)), so multiplying the stored scaling is the merge
weight — exact and reversible:
def set_lora_scale(model, w):
for mod in model.modules():
sc = getattr(mod, "scaling", None)
if isinstance(sc, dict):
if not hasattr(mod, "_base_scaling"):
mod._base_scaling = dict(sc)
for k in sc:
sc[k] = mod._base_scaling[k] * w
Every request is one `` block. The fields are fixed — none may be added or removed:
- Reference(s):
{None | Speaker: | }
- Instruction:
{GENERAL: ... and/or SCRIPT: ...}
- Tokens:
{target length in audio frames}
- Quality:
None
- Sound Event:
None
- Ambient Sound:
None
- Language:
{English | German}
- Text:
{the same script as under SCRIPT:, character for character}
| Field | What goes in it |
|---|---|
Reference(s) | `` when a reference recording of the target voice is attached, Speaker: when only a voice name is known, otherwise None. |
Instruction | A GENERAL: line, a SCRIPT: block, or both. |
Tokens | Target length in audio frames. The tokenizer runs at 12.5 frames per second, so 12.8 s = 160 frames. This is the length budget and the numbe |
Quoted from the model card on Hugging Face. The full card is behind the Hugging Face link above.