Bundle
@haoku123/dsh-voice
Full-duplex voice mode for DeepSeek Harness: streamed ASR -> LLM -> TTS with barge-in
- Source
- haoku123
- stars
- 4 stars
- License
- MIT
- Updated
- Updated yesterday
Readme
# dsh-voice
Full-duplex voice mode for DeepSeek Harness: streamed ASR → LLM → TTS with barge-in.
## Status
**v0.7.0 — press-to-talk with a live caption.**
Speak into the composer mic: the assistant silences itself (playback stops,
host synthesis queue drops), the running turn is cancelled (the stop-button
route), and your speech is transcribed by the host (SenseVoice, CN-native
simplified-Chinese ASR with punctuation + ITN) and submitted. The reply
streams back as spoken audio with live captions.
Three ways to dictate:
| gesture | behaviour |
| --- | --- |
| tap the mic | continuous dictation; the VAD segments on trailing silence |
| **hold the send key** (or the mic) | records until release, slide up to discard |
| **hold `Ctrl`** (configurable `asr.hotkey`) | same, without leaving the keyboard; `Esc` discards |
While a hold is open the overlay shows a **live caption** — the interim
transcript of what has been said so far — and keeps a spinner up after
release until the authoritative transcript lands.
Barge-in detection is triggered by the mic's leading speech edge. The ASR
engine runs an **NLMS acoustic echo canceller** (see `src/aec.ts`) using the
page's own TTS playback as the echo reference, so loud assistant audio is
subtracted from the mic before the VAD — the browser-level
`echoCancellation` constraint is kept only as a fallback when no echo
reference is available.
Live caption interims are **incremental**: each pass sends only the audio
recorded since the last pass (correlated by session header), and the host
decodes a bounded sliding window per session instead of re-decoding the
whole hold. Preview cost therefore stops growing with hold length.
## Demo

The loop: hold the composer's send key (its arrow is covered by a mic glyph),
watch the live caption fill in while you speak, release into a spinner until
the final transcript lands, then the reply streams back as spoken audio
sentence-by-sentence — until the user's voice interrupts playback and stops
the running turn mid-line (true barge-in). `Ctrl` does the same without
leaving the keyboard.
## How it works
```
input: mic ──RMS endpoint detection──▶ POST /asr (raw f32 PCM)
│ text (SenseVoice)
▼
composer draft ──submit──▶ model stream ──llm/stream tap──▶ SentenceSegmenter
│
browser ◀── SSE /dsh-voice-api/stream ── TtsQueue (msedge-tts) ◀──┘
(base64 MP3 frames + caption text)
barge-in: speech edge ──▶ engine.skip() + POST /cancel (epoch bump)
+ session.cancel() when a turn is running
```
- The `llm/stream` tap is **lossless**: every chunk is yielded unchanged, the
segmenter only observes. The model stream is never blocked by synthesis.
- ASR runs **host-side** with [sherpa-onnx](https://github.com/k2-fsa/sherpa-onnx)
(Apache-2.0) running **SenseVoice** — the CN-native speech model that
outperforms whisper on Chinese: native simplified output, punctuation,
inverse text normalization (ITN) and 50+ language auto-detection. The
browser only records and posts raw little-endian f32 PCM.
- Model files stream through a **cache-through proxy** at
`/dsh-voice-api/hf` and are mirrored to disk
(`~/.cache/dsh-voice/models/`, configurable via `cacheDir`), so every
browser/recognizer load after the first is served from local disk.
Downloads resume from partial `.part` files when interrupted. Use
`npm run prefetch` to warm the cache once.
- RMS endpoint detection: 16kHz getUserMedia, 2s trailing-silence cutoff,
max 30s segment, pre/post padding. Zero dependencies.
- **Press-to-talk bypasses the VAD entirely.** Holding the key is already the
intent, so every buffer between press and release is kept — gating on
loudness there only drops quiet speech, which is indistinguishable from a
broken button. Only captures below 250ms are discarded (mis-taps).
- **Live caption**: while a hold is open the engine re-decodes the buffer
every ~900ms and shows the interim text. SenseVoice is not a streaming
model, so this is only requested while the overlay is actually on screen,
and stops past 12s of audio. Interims are strictly previews: they never
reach the composer draft, and an interim that lands after the release is
dropped (epoch check) so it can never overwrite the final transcript.
- Barge-in is three-layered: local playback queue cleared, host `TtsQueue`
epoch bumped (queued AND in-flight synthesis dropped), and the running
turn cancelled when `session.running` is true. An aborted turn never
flushes its trailing half-sentence — exactly what the user interrupted.
- `modelHost` accepts any HF-compatible mirror (e.g. `https://hf-mirror.com`
for CN networks).
## API
| Route | Purpose |
|-------|---------|
| `GET /dsh-voice-api/stream` | SSE; `event: audio` frames `{sessionId, seq, text, audio(base64 MP3)}` |
| `POST /dsh-voice-api/asr` | raw little-endian f32 PCM body → `{text}` via SenseVoice |
| `POST /dsh-voice-api/cancel` | `{sessionId}` drops queued + in-flight synthesis (epoch bump) |
| `GET /dsh-voice-api/config` | ASR runtime config `{asr: {...}}` for the mic button |
| `GET /dsh-voice-api/hf/*` | cache-through HF model proxy (mirrors to `cacheDir`) |
| `GET /dsh-voice-api/*` | ping: `{ok, name, enabled}` |
Config (bundle patch row):
```yaml
- id: voice
name: '@haoku123/dsh-voice'
config:
voice: zh-CN-XiaoxiaoNeural
cacheDir: ~/.cache/dsh-voice/models # optional, on-disk model cache
asr:
model: csukuangfj/sherpa-onnx-sense-voice-zh-en-ja-ko-yue-2024-07-17
modelHost: https://huggingface.co # or https://hf-mirror.com
language: auto # auto | zh | en | ja | ko | yue
useItn: true # inverse text normalization
autoSend: false
mode: toggle # toggle | hold
hotkey: Control # keyboard press-to-talk; '' disables
```
Model files are fetched through the proxy on first use; warm the cache once
with the dsh host running:
```sh
npm run prefetch # uses http://127.0.0.1:3080 by default
```
## Install
```sh
dsh plugin --profile web add <repo-url-or-path>
dsh --profile web
```
Note: needs Node ≥ 22.19 or ≥ 24 (`node:zlib` zstd APIs).
## Tests
```sh
npm test # segmenter unit tests (pure, no network)
node test/host.integration.test.mjs # llm/stream tap + real Edge TTS + SSE + /config
node test/bargein.test.mjs # client inject face wiring (skipPlayback/cancelTurn)
node test/bargein-semantics.test.mjs # aborted turn no-flush + cancel drops in-flight
node verify-client.mjs # client bundle registration/exports/slots/dynamic-import
```
Install
dsh plugin --profile web add github:haoku123/dsh-voice
Profile: web
With the hub plugin installed, ask your agent to install it by name — it resolves the same plan shown here.
dsh plugin --profile web add github:stvlynn/dsh.fish#path:packages/dsh-plugin-hub
install haoku123-dsh-voice from the hub
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