Text-to-Speech
ONNX
GGUF
Chinese
English
onnxruntime
tts
on-device
jetson
telephony
vits
mb-istft-vits
multi-speaker
mandarin
taiwanese-mandarin
imatrix
conversational
Instructions to use Luigi/PrimeTTS with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Notebooks
- Google Colab
- Kaggle
- Local Apps Settings
- llama.cpp
How to use Luigi/PrimeTTS with llama.cpp:
Install (macOS, Linux)
curl -LsSf https://llama.app/install.sh | sh # Start a local OpenAI-compatible server with a web UI: llama serve -hf Luigi/PrimeTTS:F32 # Run inference directly in the terminal: llama cli -hf Luigi/PrimeTTS:F32
Install from WinGet (Windows)
winget install llama.cpp # Start a local OpenAI-compatible server with a web UI: llama serve -hf Luigi/PrimeTTS:F32 # Run inference directly in the terminal: llama cli -hf Luigi/PrimeTTS:F32
Use pre-built binary
# Download pre-built binary from: # https://github.com/ggerganov/llama.cpp/releases # Start a local OpenAI-compatible server with a web UI: ./llama-server -hf Luigi/PrimeTTS:F32 # Run inference directly in the terminal: ./llama-cli -hf Luigi/PrimeTTS:F32
Build from source code
git clone https://github.com/ggerganov/llama.cpp.git cd llama.cpp cmake -B build cmake --build build -j --target llama-server llama-cli # Start a local OpenAI-compatible server with a web UI: ./build/bin/llama-server -hf Luigi/PrimeTTS:F32 # Run inference directly in the terminal: ./build/bin/llama-cli -hf Luigi/PrimeTTS:F32
Use Docker
docker model run hf.co/Luigi/PrimeTTS:F32
- LM Studio
- Jan
- Ollama
How to use Luigi/PrimeTTS with Ollama:
ollama run hf.co/Luigi/PrimeTTS:F32
- Unsloth Studio
How to use Luigi/PrimeTTS with Unsloth Studio:
Install Unsloth Studio (macOS, Linux, WSL)
curl -fsSL https://unsloth.ai/install.sh | sh # Run unsloth studio unsloth studio -H 0.0.0.0 -p 8888 # Then open http://localhost:8888 in your browser # Search for Luigi/PrimeTTS to start chatting
Install Unsloth Studio (Windows)
irm https://unsloth.ai/install.ps1 | iex # Run unsloth studio unsloth studio -H 0.0.0.0 -p 8888 # Then open http://localhost:8888 in your browser # Search for Luigi/PrimeTTS to start chatting
Using HuggingFace Spaces for Unsloth
# No setup required # Open https://huggingface.co/spaces/unsloth/studio in your browser # Search for Luigi/PrimeTTS to start chatting
- Atomic Chat new
- Docker Model Runner
How to use Luigi/PrimeTTS with Docker Model Runner:
docker model run hf.co/Luigi/PrimeTTS:F32
- Lemonade
How to use Luigi/PrimeTTS with Lemonade:
Pull the model
# Download Lemonade from https://lemonade-server.ai/ lemonade pull Luigi/PrimeTTS:F32
Run and chat with the model
lemonade run user.PrimeTTS-F32
List all available models
lemonade list
File size: 8,122 Bytes
a37967e 1261e7b a37967e 1261e7b a37967e | 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 | #!/usr/bin/env python3
"""Export a trained zh-TW/en 8k Inflect-Nano (acoustic + snake_8k vocoder) to ONNX.
Config-driven from the train() checkpoints. FastSpeech split:
encoder.onnx -> numpy host_regulate -> decoder.onnx -> vocoder.onnx.
Validates full-pipeline parity vs torch. Run in moss-train-venv."""
from __future__ import annotations
import argparse, sys, math, json
from pathlib import Path
import numpy as np, torch
REPO = "/tmp/inflect-nano"
sys.path.insert(0, REPO)
from inflect_nano.acoustic import MicroFastSpeech, MicroFastSpeechConfig
from inflect_nano.vocoder import HifiGanGenerator, make_config
class EncoderHead(torch.nn.Module):
def __init__(self, m): super().__init__(); self.m = m
def forward(self, phone, tone, lang, speaker):
m = self.m
tok = torch.ones_like(phone, dtype=torch.bool)
enc = m.encode(phone, tone, lang, speaker, tok)
log_dur, energy, bright, pitch = m.predict_prosody(enc, tok)
dur = (torch.exp(log_dur) - 1.0).clamp(0, 80).round().clamp_min(1).long()
cond = enc + m.energy_proj(energy.unsqueeze(-1)) + m.bright_proj(bright.unsqueeze(-1))
pitch = torch.stack([pitch[..., 0], pitch[..., 1].clamp(0, 1)], dim=-1)
return cond, dur, pitch
class DecoderHead(torch.nn.Module):
def __init__(self, m): super().__init__(); self.m = m
def forward(self, frames, frame_meta, local_ctx_raw, abs_pos, pitch_frame, frame_mask):
m = self.m
x = frames + m.frame_proj(frame_meta) + m.local_ctx(local_ctx_raw)
x = x + m.abs_frame(abs_pos)
if m.cfg.use_frame_pitch:
refined = m.refine_frame_pitch(frames, frame_meta, pitch_frame)
x = x + m.pitch_proj(refined)
for blk in m.decoder:
x = blk(x, frame_mask)
x = x + m.frame_gru(x)[0]
mel = m.mel_head(x).transpose(1, 2)
return mel + m.cfg.postnet_scale * m.postnet(mel)
def host_regulate(cond, dur, pitch, abs_bins, max_frames):
c = cond[0]; d = dur[0].astype(np.int64); d[d < 0] = 0
T, H = c.shape
frames = np.repeat(c, d, axis=0); F = frames.shape[0]
tok = np.repeat(np.arange(T), d); starts = np.cumsum(d) - d
within = np.arange(F) - starts[tok]; dpf = d[tok].astype(np.float32)
rel = (within / np.maximum(dpf - 1, 1)).astype(np.float32)
tc = max(1, int((d > 0).sum())); token_pos = (tok / max(1, tc - 1)).astype(np.float32)
ld = (np.log1p(dpf) / 6.0).astype(np.float32); center = 1.0 - np.abs(rel * 2 - 1)
fm = np.stack([rel, 1 - rel, center, np.sin(rel*np.pi), np.cos(rel*np.pi), token_pos, ld, dpf/40.0], -1).astype(np.float32)
prev = np.concatenate([c[:1], c[:-1]], 0); nxt = np.concatenate([c[1:], c[-1:]], 0)
lc = np.repeat(np.concatenate([prev, c, nxt], -1), d, axis=0).astype(np.float32)
pos = np.arange(F); abs_pos = np.minimum(pos*abs_bins//max(1, max_frames), abs_bins-1).astype(np.int64)
pf = np.repeat(pitch[0], d, axis=0).astype(np.float32)
return {"frames": frames[None].astype(np.float32), "frame_meta": fm[None], "local_ctx_raw": lc[None],
"abs_pos": abs_pos[None], "pitch_frame": pf[None], "frame_mask": np.ones((1, F), bool)}
def main():
ap = argparse.ArgumentParser()
ap.add_argument("--acoustic-ckpt", required=True)
ap.add_argument("--vocoder-ckpt", required=True)
ap.add_argument("--out-dir", required=True)
ap.add_argument("--symbol-table", default="/home/luigi/jetson-tts/mossnano/zhtw8k/symbol_table.json")
args = ap.parse_args()
import onnxruntime as ort
OUT = Path(args.out_dir); OUT.mkdir(parents=True, exist_ok=True)
dev = torch.device("cpu")
ac = torch.load(args.acoustic_ckpt, map_location=dev, weights_only=False)
cfg = MicroFastSpeechConfig(**ac["config"])
m = MicroFastSpeech(cfg); m.load_state_dict(ac["model"], strict=False); m.eval()
# The group-duration planner uses a non-ONNX-able host loop and only adjusts inference-time
# durations (the mel decoder is trained on GT durations). Disable it so the exported encoder's
# plain-duration path (which keeps the contextual duration-delta) matches m.infer() for parity.
if getattr(m, "group_duration_delta", None) is not None:
m.group_duration_delta = None
print("note: group_duration_planner disabled at export (host-loop; plain durations used)")
enc, dec = EncoderHead(m).eval(), DecoderHead(m).eval()
print(f"acoustic: sr={cfg.sample_rate} vocab={cfg.vocab_size} tone={cfg.tone_size} lang={cfg.lang_size} "
f"abs_bins={cfg.abs_frame_bins} max_frames={cfg.max_frames}")
vc = torch.load(args.vocoder_ckpt, map_location=dev, weights_only=False)
vcfg = make_config(vc["config"]["variant"])
vm = HifiGanGenerator(vcfg); vm.load_state_dict(vc["generator"]); vm.remove_weight_norm(); vm.eval()
assert vcfg.sample_rate == cfg.sample_rate
# sample input: a short valid id sequence (plumbing test)
T = 40
g = torch.Generator().manual_seed(0)
phone = torch.randint(1, min(80, cfg.vocab_size), (1, T), generator=g)
tone = torch.randint(0, cfg.tone_size, (1, T), generator=g)
lang = torch.randint(0, cfg.lang_size, (1, T), generator=g)
spk = torch.zeros(1, dtype=torch.long)
with torch.no_grad():
cond, dur, pitch = enc(phone, tone, lang, spk)
reg = host_regulate(cond.numpy(), dur.numpy(), pitch.numpy(), cfg.abs_frame_bins, cfg.max_frames)
bt = tuple(torch.from_numpy(reg[k]).clone() for k in ["frames","frame_meta","local_ctx_raw","abs_pos","pitch_frame","frame_mask"])
mel_split = dec(*bt)
mel_ref = m.infer(phone, tone, lang, spk)
print(f"mel parity max_abs_diff={float((mel_ref-mel_split).abs().max()):.2e}")
torch.onnx.export(enc, (phone, tone, lang, spk), str(OUT/"acoustic_encoder.onnx"),
input_names=["phone","tone","lang","speaker"], output_names=["conditioned","durations","pitch"],
dynamic_axes={"phone":{1:"T"},"tone":{1:"T"},"lang":{1:"T"},"conditioned":{1:"T"},"durations":{1:"T"},"pitch":{1:"T"}},
opset_version=17, dynamo=False)
bn = ["frames","frame_meta","local_ctx_raw","abs_pos","pitch_frame","frame_mask"]
torch.onnx.export(dec, bt, str(OUT/"acoustic_decoder.onnx"),
input_names=bn, output_names=["mel"],
dynamic_axes={**{n:{1:"F"} for n in bn}, "mel":{2:"F"}}, opset_version=17, dynamo=False)
dummy = torch.randn(1, vcfg.num_mels, 60) # match vocoder mel count (40 for snake_8k40, 80 otherwise)
torch.onnx.export(vm, dummy, str(OUT/"vocoder.onnx"), input_names=["mel"], output_names=["wav"],
dynamic_axes={"mel":{2:"frames"},"wav":{2:"samples"}}, opset_version=17, dynamo=False)
# full-pipeline ONNX parity
sA = ort.InferenceSession(str(OUT/"acoustic_encoder.onnx"), providers=["CPUExecutionProvider"])
sB = ort.InferenceSession(str(OUT/"acoustic_decoder.onnx"), providers=["CPUExecutionProvider"])
sV = ort.InferenceSession(str(OUT/"vocoder.onnx"), providers=["CPUExecutionProvider"])
oc, od, op = sA.run(None, {"phone":phone.numpy(),"tone":tone.numpy(),"lang":lang.numpy(),"speaker":spk.numpy()})
reg2 = host_regulate(oc, od, op, cfg.abs_frame_bins, cfg.max_frames)
feeds = {n:(reg2[n].astype(np.float32) if reg2[n].dtype!=bool else reg2[n]) for n in bn}
feeds["abs_pos"] = reg2["abs_pos"].astype(np.int64)
mel_onnx = sB.run(None, feeds)[0]
wav_onnx = sV.run(None, {"mel": mel_onnx.astype(np.float32)})[0]
with torch.inference_mode(): wav_ref = vm(mel_ref).numpy()
n = min(wav_ref.shape[-1], wav_onnx.shape[-1])
print(f"FULL-PIPELINE wav parity max_abs_diff={float(np.abs(wav_ref[...,:n]-wav_onnx[...,:n]).max()):.2e}")
# save metadata for the Nano runtime
json.dump({"sample_rate":cfg.sample_rate,"abs_frame_bins":cfg.abs_frame_bins,"max_frames":cfg.max_frames,
"hop_size":vcfg.hop_size,"n_mels":cfg.n_mels,"use_frame_pitch":cfg.use_frame_pitch},
open(OUT/"meta.json","w"), indent=1)
print("sizes(KB):", {f.name: f.stat().st_size//1024 for f in OUT.glob("*.onnx")})
print("EXPORT_OK", OUT)
if __name__ == "__main__":
main()
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