dsh-voice
martindye/dsh-voice
Two-way voice for the DSH web app: dictate with the unified composer, hear replies with speaker buttons and auto-read, on any device. Needs DSH 0.1.7+, Node 22+, Python 3 with numpy and requests, a TTS model, and a microphone.
Installation
dsh plugin --profile web add github:martindye/dsh-voiceREADME
dsh-voice
Martin's self-contained two-way voice plugin for the DSH web app: you speak, the agent replies out loud, and the microphone keeps listening. Works on any device the web app runs on (iPhone or PC).
A standalone fork of @deepseek-ai/dsh-experimental-client-ui-voice-input
(0.1.7-rc.2, from the deepseek-harness checkout), renamed @martindye/dsh-voice
and grown into the single voice plugin: composer in, speaker buttons and
auto-read out. The retired dsh-voice-chat kit and the old dsh-voice-output
plugin are both superseded by this repository.
What it does
Input (the composer):
- The unified 34 px blue composer button: voice entry on an empty draft, send arrow while typing, stop square while the agent runs or the system speaks, cross while a voice turn is active.
- Continuous listening: after a reply ends the microphone reopens on its own; an auto-open that draws no words is quietly re-offered a few times.
- The spoken pause command ("hold the mic", "pause the mic", "stop listening") parks the listening loop without sending.
- In-call mute: while a voice turn is active the dictate button becomes a mute/unmute toggle (white on red while muted); muted audio never reaches the recognizer and the voice-activity clocks freeze.
Output (the speaker):
- A speaker button on every finalized assistant message (speak / stop).
- Auto-read: fresh replies (younger than 2 minutes) are read aloud on a
device that (1) has the auto-read toggle on and (2) has used the voice
composer on that device (
dsh.voice-chat.v1). No PC-side bridge is involved — each device reads what it is shown. - A per-device auto-read toggle in the session header corner (green dot when the device is voice-armed), persisted per browser.
- Text is resolved from the session log on the host and prepared for speech (code, links and markdown stripped) before synthesis.
Layout
src/— plugin source.src/index.ts(host entry),src/tts-host.ts(TTS supervision + speech preparation, host half),src/client/(browser half: composer, speaker button, header toggle, playback).tts/— the plugin's own TTS HTTP server (server.py) and its engines (qwen3.pyGPU voice,pocket.pyCPU fallback).bin/tts-server.exe— the bundled qwentts.cpp server for the qwen3 engine (adopts a compatible resident server on 127.0.0.1:8095 when present).ref_voices/— the reference voices the qwen3 engine clones from (pocket,warm-brit). Both are synthetic takes (Pocket TTS and a VoiceDesign take respectively) — no real person's voice is bundled; seeTHIRD-PARTY.mdfor the full attribution list.lib/— build output (lib/index.jsnode half,lib/client.jsbrowser bundle). Gitignored; rebuild withpnpm bundle.tests/— vitest client tests (jsdom): the composer audio pipeline, the waveform, and the conversation simulation with its trace files.
Building
pnpm install
pnpm bundle # lib/index.js + lib/client.js
pnpm test # vitest
The client bundle is a CJS closure factory registered with the host's module loader under the package name; platform modules (react, primitives, slots, gateway) stay externals, everything else inlines. CSS modules are compiled with lightningcss and injected as self-managed style tags.
Runtime requirements (the machine running DSH)
- Python 3 on PATH (override with
DSH_VOICE_PYTHON) withnumpyandrequests; addpocket-ttsfor the CPU fallback engine. The server is spawned on demand at127.0.0.1:8189(override the port withDSH_VOICE_TTS_PORT) and supervised by the plugin host. - qwen3 (GPU) models under
~/.dsh/voice-chat/models/qwen3(talker + tokenizer GGUFs); the engine choice is the shared consent file~/.dsh/voice-chat/consent.json("engine": "qwen3" | "pocket"), default qwen3 when its models are present. - Host log:
~/.dsh/logs/voice-tts.log.
With Pocket selected, only the CPU engine is warmed when the TTS server starts.
A failed Pocket request does not fall back to a GPU engine. Selecting qwen3
still enables the GPU engine and retains its CPU fallback. The host passes the
selected registry name through VOICE_CHAT_PRELOAD_ENGINE; a standalone server
or an older host defaults its warm-up to Pocket. GPU requests remain available
on demand. Run python -m unittest discover -s tests -p 'test_tts_server.py'
for the model-free startup and fallback checks.
Installing
From npm (the published package ships prebuilt lib/ — no build step, no
build-script permission needed):
dsh plugin --profile <profile> add @martindye/dsh-voice
for example dsh plugin --profile web add @martindye/dsh-voice for the web
app. Then restart the web host (or start it) and open the app: the unified
voice button appears in the composer, and a speaker button on every finalized
message.
Requirements: DSH 0.1.7+ (web app), Node 22+ on the host, and Python 3 with
the TTS engine dependencies for the spoken replies (see tts/); the browser
needs microphone permission for dictation, and the speech-to-text models are
fetched on first use under the DSH home.
How the bundle works
The package declares a dsh.bundle with cordis.patch.yml. Adding the
bundle to a profile's bundles list (the web profile does this in its
package.json) inserts:
ui-voice— this plugin,- the speech-to-text stack (
speech-to-textwithsensevoice-localas default provider and languageen,speech-to-text-sensevoicewithdataRootat the DSH homespeech-to-text/sensevoice, andapi-speech-to-text),
and disables the upstream ui-voice-input row. The three @deepseek-ai/ dsh-experimental-*speech-to-text* packages must be installed in the same
profile (the web profile links them from the deepseek-harness checkout).
lib/ is not committed to git (it is included in the npm tarball); run
pnpm bundle after pulling when working from a checkout. A page refresh
picks up the client; the host half needs a web-host restart.
Changelog (the latest updates, by version, as of 24/10)
1.0.6 (24/10)
- The changelog now carries the version and publish date per entry.
1.0.5 (24/10)
- Two new settings in the settings card: the auto-read freshness cutoff (how old a reply may be and still be read on auto, 30 s to 10 min) and the pause that ends dictation (1 to 10 seconds of silence, defaulting to 4 s).
- A conversation switch no longer kills the voice conversation: coming back restores the row where it left off (armed, muted, parked) and the microphone reopens without a tap.
1.0.2 (24/10)
- A run that opened muted keeps its webm header, so the later unmuted words decode instead of failing with "unable to decode audio".
- One microphone in the error row: the mute leaves the retry row and appears only where a mic is actually open.
1.0.1 (22/10)
- One cross, one job: the cross is the single stop control in every phase — it stops the turn, the speech and the words in flight.
- A dead input device is named when a recording draws no audio ("check the input device"), instead of a generic retry.
- Auto-reads queue behind audio that is already playing instead of cutting it off; a manual read or an explicit stop still cuts in.
- Dictated messages can carry a flag (a marker line or a sidecar note) so they are recognisable in the log; typed messages never do.
- The microphone stays open while the agent thinks, so words spoken mid-turn are transcribed and queued.
- Voice activity trusts the recorder only when the analyser is blind (a suspended page), so a quiet room never reads as speech.
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