Drop audioop: request 16 kHz directly via Speaches sample_rate extension

Settles the Python version question: floor >=3.11, no ceiling.
Container moves to python:3.13-slim, CI tests on 3.13. The embedded
Piper fallback resamples with numpy (already present via [speech]).
Verified with a live round trip at 16 kHz.

Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>

.gitea/workflows/build.yml

@@ -15,7 +15,7 @@ jobs:
 
       - uses: actions/setup-python@v5
         with:
-          python-version: '3.12'
+          python-version: '3.13'
 
       - name: Install
         run: pip install -e "./gateway[dev]"

AGENTS.md

@@ -15,9 +15,11 @@ one repo:
   (Tatlock `/v1/chat/completions`) → TTS. Listens on port **8600**. STT/TTS models live
   in the shared **Speaches** container (live on port 8601, OpenAI-format API), not in
   the gateway image; `stt.py`/`tts.py` are pluggable backends (`speaches` default,
-  `embedded` fallback needing the `[speech]` extra). Gateway runs on **Python 3.12
-  exactly** — system python3 on tower-of-joy is 3.8, and `audioop` (used for TTS
-  resampling) is removed in 3.13.
+  `embedded` fallback needing the `[speech]` extra). Gateway needs **Python ≥ 3.11**
+  (no ceiling; the container runs 3.13) — but system python3 on tower-of-joy is 3.8,
+  so `make setup` explicitly uses `python3.12`. No local audio resampling in the
+  default path: the gateway requests 16 kHz output via Speaches' `sample_rate`
+  extension (verified live).
 
 The device and gateway speak a WebSocket protocol defined in `docs/architecture.md`.
 **That doc is the contract** — update it in the same change as any protocol edit on

docs/architecture.md

@@ -98,8 +98,9 @@ we may adopt later for streaming transcription.
 
 - **Deployed 2026-07-14**: `ghcr.io/speaches-ai/speaches:latest-cuda` on host port
   **8601**, with `Systran/faster-whisper-small` (STT) and
-  `speaches-ai/Kokoro-82M-v1.0-ONNX` (TTS, 24 kHz — the gateway resamples to the
-  16 kHz device contract; default voice `bm_george`, en-GB male). LAN-only like the
+  `speaches-ai/Kokoro-82M-v1.0-ONNX` (TTS — Kokoro is natively 24 kHz, but the
+  gateway requests the 16 kHz device contract directly via Speaches' `sample_rate`
+  extension, verified live; default voice `bm_george`, en-GB male). LAN-only like the
   Tatlock internal route — do not expose through NPM without auth. Register in
   `CONTAINERS.md`.
 - **Measured** (live round trip through the gateway code, warm): STT ~0.3 s for a

gateway/Dockerfile

@@ -1,4 +1,4 @@
-FROM python:3.12-slim
+FROM python:3.13-slim
 
 WORKDIR /app
 

gateway/Makefile

@@ -1,6 +1,6 @@
 .PHONY: setup run test lint typecheck clean
 
-# audioop pins us below 3.13; system python3 on tower-of-joy is 3.8
+# any Python >= 3.11 works; system python3 on tower-of-joy is 3.8, hence explicit
 PYTHON ?= python3.12
 
 setup:

gateway/src/desklock_gateway/tts.py

@@ -1,11 +1,10 @@
 """Text-to-speech: Speaches/Kokoro over HTTP (default) or embedded Piper.
 
 Output is always s16le mono PCM at settings.sample_rate (the device WS contract).
-Kokoro synthesizes at 24 kHz, so the speaches path resamples via audioop —
-which pins the runtime to Python 3.12 (audioop is removed in 3.13).
+The speaches path requests the target rate directly via the server's
+`sample_rate` extension, so no local resampling is needed.
 """
 
-import audioop
 import io
 import wave
 
@@ -35,6 +34,7 @@ def _synthesize_speaches(text: str) -> bytes:
             "voice": settings.tts_voice,
             "input": text,
             "response_format": "wav",
+            "sample_rate": settings.sample_rate,
         },
     )
     response.raise_for_status()
@@ -42,12 +42,12 @@ def _synthesize_speaches(text: str) -> bytes:
     with wave.open(io.BytesIO(response.content), "rb") as w:
         rate, channels, width = w.getframerate(), w.getnchannels(), w.getsampwidth()
         frames = w.readframes(w.getnframes())
-    if width != 2:
-        frames = audioop.lin2lin(frames, width, 2)
-    if channels == 2:
-        frames = audioop.tomono(frames, 2, 0.5, 0.5)
-    if rate != settings.sample_rate:
-        frames, _state = audioop.ratecv(frames, 2, 1, rate, settings.sample_rate, None)
+    if (rate, channels, width) != (settings.sample_rate, 1, 2):
+        raise RuntimeError(
+            f"Speaches returned {rate} Hz/{channels}ch/{8 * width}-bit audio; "
+            f"expected {settings.sample_rate} Hz mono 16-bit — check the server's "
+            "sample_rate support"
+        )
     return frames
 
 
@@ -61,4 +61,14 @@ def _synthesize_embedded(text: str) -> bytes:
     chunks = bytearray()
     for chunk in _embedded_voice.synthesize_stream_raw(text):
         chunks.extend(chunk)
-    return bytes(chunks)
+    native_rate = _embedded_voice.config.sample_rate
+    if native_rate == settings.sample_rate:
+        return bytes(chunks)
+
+    import numpy as np
+
+    audio = np.frombuffer(bytes(chunks), dtype=np.int16)
+    n_out = int(len(audio) * settings.sample_rate / native_rate)
+    positions = np.linspace(0, len(audio) - 1, n_out)
+    resampled = np.interp(positions, np.arange(len(audio)), audio.astype(np.float32))
+    return resampled.astype(np.int16).tobytes()