commit c2b92d96cb74611c14f111dcb6351958ac4cd17c
Author: Brian Hetherman <bhetherman@gmail.com>
Date:   Mon Aug 31 00:47:35 2026 -0400

    Initial voice-assistant-stack: open-webui + spotify-voice-assistant submodules, asr/tts sidecars
    
    - open-webui: submodule, LAN OIDC/Authentik-fronted, GPU-passthrough ollama,
      ollama-auth proxy, asr/tts wired via docker-compose.audio.yaml
    - spotify-voice-assistant: submodule, Home Assistant custom integration
    - tts-server / asr-server: plain directories, built as sidecars by
      open-webui/docker-stack.sh (no separate git history)

diff --git a/.gitignore b/.gitignore
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--- /dev/null
+++ b/.gitignore
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+tts-server/audio-files/
diff --git a/.gitmodules b/.gitmodules
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--- /dev/null
+++ b/.gitmodules
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+[submodule "open-webui"]
+	path = open-webui
+	url = ssh://git@192.168.50.224:30009/bhetherman/open-webui.git
+[submodule "spotify-voice-assistant"]
+	path = spotify-voice-assistant
+	url = https://git.hetherman.cloud/bhetherman/spotify-voice-assistant.git
diff --git a/asr-server/Dockerfile b/asr-server/Dockerfile
new file mode 100644
index 0000000..917b1f9
--- /dev/null
+++ b/asr-server/Dockerfile
@@ -0,0 +1,17 @@
+FROM python:3.11-slim
+
+RUN apt-get update && apt-get install -y --no-install-recommends \
+    libsndfile1 ffmpeg \
+    && rm -rf /var/lib/apt/lists/*
+
+WORKDIR /app
+# The Titan Xp (Pascal, sm_61) needs a torch build old enough to still ship
+# Pascal kernels; recent default PyPI wheels dropped sm_60/sm_61 entirely.
+ENV PIP_EXTRA_INDEX_URL=https://download.pytorch.org/whl/cu124
+COPY requirements.txt .
+RUN pip install --no-cache-dir -r requirements.txt
+
+COPY app.py .
+
+EXPOSE 8020
+CMD ["uvicorn", "app:app", "--host", "0.0.0.0", "--port", "8020"]
diff --git a/asr-server/app.py b/asr-server/app.py
new file mode 100644
index 0000000..567ffc0
--- /dev/null
+++ b/asr-server/app.py
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+import hmac
+import logging
+import os
+import tempfile
+
+import torch
+from fastapi import Depends, FastAPI, File, Form, HTTPException, UploadFile
+from fastapi.responses import JSONResponse, PlainTextResponse
+from fastapi.security import HTTPAuthorizationCredentials, HTTPBearer
+from transformers import AutoModelForRNNT, AutoProcessor
+from transformers.audio_utils import load_audio
+
+logging.basicConfig(level=logging.INFO)
+log = logging.getLogger("asr-server")
+
+AUTH_TOKEN = os.environ["ASR_AUTH_TOKEN"]
+bearer_scheme = HTTPBearer()
+
+
+def require_auth(creds: HTTPAuthorizationCredentials = Depends(bearer_scheme)) -> None:
+    if not hmac.compare_digest(creds.credentials, AUTH_TOKEN):
+        raise HTTPException(status_code=401, detail="invalid token")
+
+
+MODEL_ID = os.environ.get("ASR_MODEL_ID", "nvidia/nemotron-3.5-asr-streaming-0.6b")
+DEVICE = "cuda" if torch.cuda.is_available() else "cpu"
+DTYPE = torch.float16 if DEVICE == "cuda" else torch.float32
+
+# The RNN-T decoder's LSTM layers hit cuDNN's fused RNN kernel on `.to(device)`,
+# which requires SM >= 7.5 in recent cuDNN builds. This GPU (Pascal, SM 6.1) is
+# below that floor, so disable cuDNN and fall back to PyTorch's generic CUDA
+# RNN kernels instead.
+torch.backends.cudnn.enabled = False
+
+app = FastAPI()
+
+log.info("Loading %s on %s (%s)", MODEL_ID, DEVICE, DTYPE)
+processor = AutoProcessor.from_pretrained(MODEL_ID)
+model = AutoModelForRNNT.from_pretrained(MODEL_ID, dtype=DTYPE).to(DEVICE).eval()
+SAMPLING_RATE = processor.feature_extractor.sampling_rate
+log.info("Model ready (sampling_rate=%s)", SAMPLING_RATE)
+
+
+@app.get("/health")
+def health():
+    return {"status": "ok", "device": DEVICE}
+
+
+@app.post("/v1/audio/transcriptions", dependencies=[Depends(require_auth)])
+async def transcribe(
+    file: UploadFile = File(...),
+    model_name: str = Form("nemotron-3.5-asr-streaming-0.6b", alias="model"),
+    language: str | None = Form(None),
+    response_format: str = Form("json"),
+):
+    suffix = os.path.splitext(file.filename or "")[1] or ".wav"
+    data = await file.read()
+
+    with tempfile.NamedTemporaryFile(suffix=suffix) as tmp:
+        tmp.write(data)
+        tmp.flush()
+        try:
+            audio = load_audio(tmp.name, sampling_rate=SAMPLING_RATE)
+        except Exception as exc:
+            raise HTTPException(status_code=400, detail=f"could not decode audio: {exc}") from exc
+
+    lang = language or "auto"
+    inputs = processor(audio, sampling_rate=SAMPLING_RATE, language=lang)
+    inputs = inputs.to(DEVICE, dtype=DTYPE)
+
+    with torch.inference_mode():
+        output = model.generate(**inputs, return_dict_in_generate=True)
+
+    text = processor.decode(output.sequences, skip_special_tokens=True)
+    if isinstance(text, list):
+        text = text[0] if text else ""
+
+    if response_format == "text":
+        return PlainTextResponse(text)
+    return JSONResponse({"text": text})
diff --git a/asr-server/requirements.txt b/asr-server/requirements.txt
new file mode 100644
index 0000000..110f592
--- /dev/null
+++ b/asr-server/requirements.txt
@@ -0,0 +1,8 @@
+fastapi==0.115.*
+uvicorn[standard]==0.32.*
+python-multipart==0.0.*
+torch==2.6.0
+transformers>=5.13.0
+soundfile
+librosa
+accelerate
diff --git a/assistant-stack.code-workspace b/assistant-stack.code-workspace
new file mode 100644
index 0000000..c819b0a
--- /dev/null
+++ b/assistant-stack.code-workspace
@@ -0,0 +1,20 @@
+{
+	"folders": [
+		{
+			"path": "open-webui"
+		},
+		{
+			"path": ".venv"
+		},
+		{
+			"path": "tts-server"
+		},
+		{
+			"path": "asr-server"
+		},
+		{
+			"path": "spotify-voice-assistant"
+		}
+	],
+	"settings": {}
+}
\ No newline at end of file
diff --git a/open-webui b/open-webui
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index 0000000..7eff527
--- /dev/null
+++ b/open-webui
@@ -0,0 +1 @@
+Subproject commit 7eff527d16f2a143c28bd04f51cb388a7055966d
diff --git a/spotify-voice-assistant b/spotify-voice-assistant
new file mode 160000
index 0000000..7249919
--- /dev/null
+++ b/spotify-voice-assistant
@@ -0,0 +1 @@
+Subproject commit 72499190068f3db057d55013a45846bdccf11fc8
diff --git a/tts-server/Dockerfile b/tts-server/Dockerfile
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index 0000000..43166dc
--- /dev/null
+++ b/tts-server/Dockerfile
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+FROM python:3.11-slim
+
+RUN apt-get update && apt-get install -y --no-install-recommends \
+    libsndfile1 ffmpeg espeak-ng \
+    && rm -rf /var/lib/apt/lists/*
+
+WORKDIR /app
+# The Titan Xp (Pascal, sm_61) needs a torch build old enough to still ship
+# Pascal kernels; recent default PyPI wheels dropped sm_60/sm_61 entirely.
+ENV PIP_EXTRA_INDEX_URL=https://download.pytorch.org/whl/cu121
+COPY requirements.txt .
+RUN pip install --no-cache-dir -r requirements.txt
+
+COPY app.py .
+
+EXPOSE 8010
+CMD ["uvicorn", "app:app", "--host", "0.0.0.0", "--port", "8010"]
diff --git a/tts-server/app.py b/tts-server/app.py
new file mode 100644
index 0000000..27a6dc9
--- /dev/null
+++ b/tts-server/app.py
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+import asyncio
+import hmac
+import io
+import logging
+import os
+from concurrent.futures import ThreadPoolExecutor
+from pathlib import Path
+
+import numpy as np
+import parselmouth
+import soundfile as sf
+import torch
+from fastapi import Depends, FastAPI, HTTPException
+from fastapi.responses import Response
+from fastapi.security import HTTPAuthorizationCredentials, HTTPBearer
+from kokoro import KPipeline
+from pydantic import BaseModel
+from sopro import SoproTTS
+
+logging.basicConfig(level=logging.INFO)
+log = logging.getLogger("tts-server")
+
+AUTH_TOKEN = os.environ["TTS_AUTH_TOKEN"]
+bearer_scheme = HTTPBearer()
+
+
+def require_auth(creds: HTTPAuthorizationCredentials = Depends(bearer_scheme)) -> None:
+    if not hmac.compare_digest(creds.credentials, AUTH_TOKEN):
+        raise HTTPException(status_code=401, detail="invalid token")
+
+
+# 'a' = American English. Kokoro's voices are fixed pretrained presets (not
+# zero-shot cloning), so there's no per-request drift to work around like
+# Audio8 needed -- picking a voice is just picking an ID.
+LANG_CODE = os.environ.get("TTS_LANG_CODE", "a")
+DEFAULT_VOICE = os.environ.get("TTS_DEFAULT_VOICE", "af_heart")
+KOKORO_SAMPLE_RATE = 24000
+
+# Sopro is zero-shot voice cloning, not preset voices -- a "voice" here is
+# just a reference clip dropped into this directory as <name>.wav, picked by
+# the `voice` request field.
+VOICES_DIR = Path(os.environ.get("TTS_VOICES_DIR", "/app/voices"))
+DEFAULT_SOPRO_VOICE = os.environ.get("TTS_SOPRO_DEFAULT_VOICE")
+
+DEVICE = "cuda" if torch.cuda.is_available() else "cpu"
+
+app = FastAPI()
+
+log.info("Loading Kokoro (lang_code=%s) on %s", LANG_CODE, DEVICE)
+kokoro_pipeline = KPipeline(lang_code=LANG_CODE, device=DEVICE)
+log.info("Kokoro ready")
+
+log.info("Loading Sopro on %s", DEVICE)
+sopro_model = SoproTTS.from_pretrained("samuel-vitorino/sopro-v2-turbo", device=DEVICE)
+log.info("Sopro ready")
+
+# References are just resampled/cropped tensors of the voice clip -- cheap to
+# keep around per voice name instead of redoing that work every request.
+_sopro_ref_cache = {}
+
+# Sopro caches a CUDA graph per decode length (sopro/nn/decode.py) captured on
+# whichever thread first hits that length. Replaying -- or sampling after
+# replaying -- from a different thread breaks torch's CUDA RNG bookkeeping
+# ("Offset increment outside graph capture encountered unexpectedly"), so
+# every Sopro call has to run on this same dedicated thread.
+_sopro_executor = ThreadPoolExecutor(max_workers=1)
+
+
+def _to_numpy(audio) -> np.ndarray:
+    if isinstance(audio, torch.Tensor):
+        return audio.detach().cpu().numpy()
+    return audio
+
+
+def _time_stretch(audio: np.ndarray, sample_rate: int, speed: float) -> np.ndarray:
+    # Sopro has no native speed/duration control, unlike Kokoro's duration
+    # predictor. PSOLA (pitch-synchronous overlap-add) changes tempo without
+    # pitch-shifting, and handles speech transients/consonants far more
+    # cleanly than a generic STFT phase vocoder does.
+    snd = parselmouth.Sound(audio.astype(np.float64), sampling_frequency=sample_rate)
+    stretched = parselmouth.praat.call(snd, "Lengthen (overlap-add)", 75, 600, 1.0 / speed)
+    return stretched.values[0].astype(np.float32)
+
+
+def _sopro_reference(name: str):
+    if name not in _sopro_ref_cache:
+        wav_path = VOICES_DIR / f"{name}.wav"
+        if not wav_path.exists():
+            raise HTTPException(status_code=400, detail=f"unknown sopro voice '{name}' (expected {wav_path})")
+        _sopro_ref_cache[name] = sopro_model.prepare_reference(ref_audio_path=str(wav_path))
+    return _sopro_ref_cache[name]
+
+
+def _synthesize_kokoro(text: str, voice: str, speed: float) -> tuple[np.ndarray, int]:
+    # KPipeline yields one Result per chunk it splits the input into -- no
+    # manual chunking needed here, unlike the autoregressive model this
+    # replaced.
+    chunks = [_to_numpy(result.audio) for result in kokoro_pipeline(text, voice=voice, speed=speed)]
+    audio = np.concatenate(chunks) if len(chunks) > 1 else chunks[0]
+    return audio, KOKORO_SAMPLE_RATE
+
+
+def _synthesize_sopro(text: str, voice: str | None, speed: float) -> tuple[np.ndarray, int]:
+    voice = voice or DEFAULT_SOPRO_VOICE
+    if not voice:
+        raise HTTPException(status_code=400, detail="no voice given and TTS_SOPRO_DEFAULT_VOICE is unset")
+    ref = _sopro_reference(voice)
+    audio = _to_numpy(sopro_model.synthesize(text, ref=ref))
+    if speed != 1.0:
+        audio = _time_stretch(audio, sopro_model.sample_rate, speed)
+    return audio, sopro_model.sample_rate
+
+
+class SpeechRequest(BaseModel):
+    model: str | None = None
+    input: str
+    voice: str | None = None
+    response_format: str = "wav"
+    speed: float | None = None
+
+
+@app.get("/health")
+def health():
+    return {"status": "ok", "device": DEVICE}
+
+
+@app.post("/v1/audio/speech", dependencies=[Depends(require_auth)])
+async def synthesize(req: SpeechRequest):
+    if (req.model or "").lower().startswith("sopro"):
+        loop = asyncio.get_running_loop()
+        audio, sample_rate = await loop.run_in_executor(
+            _sopro_executor, _synthesize_sopro, req.input, req.voice, req.speed or 1.0
+        )
+    else:
+        audio, sample_rate = await asyncio.to_thread(
+            _synthesize_kokoro, req.input, req.voice or DEFAULT_VOICE, req.speed or 1.0
+        )
+
+    buf = io.BytesIO()
+    fmt = "WAV" if req.response_format in ("wav", None) else req.response_format.upper()
+    sf.write(buf, audio, sample_rate, format=fmt)
+    buf.seek(0)
+
+    media_type = "audio/wav" if fmt == "WAV" else f"audio/{req.response_format}"
+    return Response(content=buf.read(), media_type=media_type)
diff --git a/tts-server/requirements.txt b/tts-server/requirements.txt
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+fastapi==0.115.*
+uvicorn[standard]==0.32.*
+torch==2.5.1
+# Pinned to match torch==2.5.1+cu121 above -- an unpinned install pulls a
+# newer torchaudio built against a CUDA runtime the Titan Xp's pinned torch
+# doesn't ship, which fails with "libcudart.so.13: cannot open shared object".
+torchaudio==2.5.1
+kokoro>=0.9.2
+sopro
+soundfile
+praat-parselmouth
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