- is_tts_echo sliding window: return True immediately when ratio >= threshold
instead of scanning all remaining windows. Common echo case drops from
~4s to <1ms for long spoken text (found by /simplify-code efficiency review).
- Clear _voice_barge_phase in _voice_submit_barge_utterance finally block
alongside _voice_barge_capture, preventing stale phase from a previous
trip affecting a future call.
- Add _voice_last_tts_text and _voice_barge_phase to _make_voice_cli test
helper so it matches __init__ state.
Reviewer feedback on the fragment-echo fallback added in 24730b10c:
- A short playback-phase transcript (e.g. a genuine one-word "yes")
could trivially match a same-length window of a longer spoken reply
at ratio 1.0 and be wrongly dropped as a self-capture. The fallback
now requires the transcript to be at least
MIN_FRAGMENT_LENGTH_FOR_ECHO characters before it runs.
- The fallback split on whitespace, so it never engaged for
no-whitespace languages (both transcript and spoken text collapse to
a single "word"). Switched the sliding window to be character-based
instead of word-based, matching the function's tokenization-independent
contract.
Adds regression tests for both cases.
is_tts_echo() compared the captured transcript against the *entire*
spoken text with a whole-string similarity ratio, which only scores high
when the two strings are close in length. But a playback-phase barge
capture is cut immediately when the trigger fires and only spans the
pre-roll buffer plus time-to-silence, so a genuine self-capture is
typically a short fragment of a longer reply, not a near-verbatim repeat
of the whole thing -- for any response longer than a clause, the
length-diluted ratio fell below threshold and the echo sailed through
ungated.
When the whole-string check misses, also slide a window sized to the
transcript's word count across the spoken text and compare against each
window, so a short fragment echoed from within a much longer
multi-sentence reply is still caught. Add regression tests for a short
fragment matched at the start and in the middle of a longer reply.
The full-duplex barge-in listener added in 5081551f0 stays active during
TTS playback with no acoustic echo cancellation. On some speaker/mic
combinations, TTS bleed alone crosses the barge threshold, gets
transcribed, and is queued as the next user turn -- whose reply is then
spoken, captured, and queued again, producing an unbounded TTS -> STT ->
TTS feedback loop (#75780).
Add a fail-closed transcript-level guard: when a barge trip happens during
the playback phase, compare the captured transcript against the TTS text
Hermes just spoke (tools/voice_mode.is_tts_echo, a language-agnostic
character-level similarity ratio). A close match is dropped instead of
queued, and the mic is handed back to the normal continuous-listening
loop. Generation-phase trips (no TTS playing, so no bleed is possible)
are unaffected.
Replaces the half-duplex per-playback barge monitors with ONE listener
that runs for the entire agent turn in continuous voice mode: armed at
utterance-submit, disarmed when the turn is fully done (response + TTS
finished). Fixes Teknium's live report that voice interruption never
works: (a) not while the LLM is generating, (b) not while TTS plays.
Root causes:
- HALF-DUPLEX GAP: the barge monitor only spawned when TTS playback
STARTED (cli.py streaming/whole-file paths, gateway _tts_stream_begin).
During LLM generation there was NO microphone listener at all.
- PLAYBACK DEAFNESS: the monitor calibrated its VAD noise floor WHILE the
speaker was blasting TTS (speaker bleed baked into the floor), then
multiplied it by 8x with a 1s strictly-consecutive block requirement —
normal speech could rarely reach the trigger, and the 2s grace
swallowed early interjections.
New model — tools/voice_mode.full_duplex_listen():
- Pre-playback calibration: quiet-room noise floor established at turn
start and HELD through playback (never recalibrated against bleed).
- Phase-aware trigger: generation = floor x voice.barge_in_threshold_multiplier
(new config, default 3.0, justified by synthetic-frame tests);
playback = additionally clamped to a 1500-RMS minimum so bleed alone
can't trip; 4000-RMS ceiling keeps speech always reachable.
- Windowed-majority detection (>=80% of a 300ms window) instead of the
strictly-consecutive counter that reset on intra-word energy dips.
- Grace on playback ONSET only (voice.barge_in_grace_seconds, default
down 2.0 -> 0.5) — suppresses the onset transient, not the mic.
- Debug diagnostics at every decision point (calibrated floor, per-window
RMS above 50% of trigger, trip/no-trip, grace suppressions) — always
logger.debug, mirrored to stderr under HERMES_VOICE_DEBUG=1.
Phase behavior (CLI cli.py + tui_gateway/server.py, same model):
- generation: speech interrupts the in-flight turn via the SAME seam the
typed/Ctrl+C interrupt uses (agent.interrupt()), cuts any pending TTS
pipeline so the stale reply never plays, and submits the captured
interjection (pre-roll capture, first syllable kept) as the next turn.
- playback: cuts TTS (streaming pipeline stop + fallback speak stop
events + file player) and submits the capture.
- stop phrase honored in BOTH phases: mid-generation 'stop' interrupts
the turn AND ends the voice chat (stop everything).
- one listener instance spans generation -> playback (no re-arm race);
double-arm refused (CLI _voice_fd_active / gateway _fd_listener_active).
Gateway specifics: _arm_full_duplex_listener() at _run_prompt_submit turn
start and inside _tts_stream_begin; _speak_text_with_barge registers its
(stop, done) pair in _fd_speak_pipelines so fallback speaks are cut and
tracked; _tts_stream_barge_in_monitor kept as a shim that arms the new
listener. Desktop renderer owns its own mic path (voice-barge-in.ts) and
is unaffected; if desktop backend-mic mode is used it inherits via the
gateway.
Tests: full_duplex_listen synthetic-RMS suite (speech-over-bleed trips,
bleed alone doesn't, quiet floor held through playback, grace window,
multiplier math 3x vs 8x, windowed-majority dips), CLI listener phase
tests (generation interrupt seam, playback cut, lifecycle spans phases,
double-arm, config forwarding, stop-phrase-mid-generation), gateway
generation-phase interrupt + stop-phrase tests. Generation-interrupt
test sabotage-verified.
Long thinking/tool stretches in a voice conversation are dead air — the
user cannot tell whether the agent is alive. New: quiet, repeating soft
bubble blips while the agent works and no speech audio is flowing.
- tools/voice_mode.py: numpy-synthesized blips (no binary assets) — two
alternating low pitches (G4/E4) with pitch glide + smooth attack/decay
envelopes, ~0.8-1.2s randomized spacing, volume = voice.beep_volume * 0.5.
start_thinking_sound(should_play=...) / stop_thinking_sound() daemon-loop
lifecycle; macOS-TCC-safe (sounddevice output gated there → silent skip,
no per-second afplay churn). New mark_audio_output_active()/
is_audio_output_active() ref-count wraps play_audio_file and the
streaming OutputStream sentence writes so "audio is flowing" is accurate.
- Config: voice.thinking_sound (default true) off-switch.
- cli.py: starts when a voice-mode turn begins, per-blip gate skips while
TTS speaks / mic records / barge capture owns the mic; stopped in the
chat() finally on every exit path.
- tui_gateway/server.py: same lifecycle around _run_prompt_submit turns
(voice mode on), gated on is_audio_output_active + continuous capture.
- Desktop: renderer owns voice-conversation audio, so a matching WebAudio
implementation (src/lib/thinking-sound.ts, same envelope/pitches) runs
while conversation status === "thinking"; honors voice.thinking_sound
(via config store) and the shared sound-mute toggle; stops instantly on
speaking/listening/end.
One owner for the wording: voice_stop_hint() in tools/voice_mode.py —
sources the phrase from voice.stop_phrases (first entry) so a custom
phrase renders correctly, and returns "" when the feature is disabled
(stop_phrases: []) so no surface shows a hint.
- CLI: printed in /voice on output (style-matched dim notice).
- TUI: voice.toggle action=on now carries stop_hint; the Ink client
renders it in the "Voice mode enabled" block (older gateways omit
the field — no hint, no crash).
- Desktop: the renderer voice loop never touches tools/voice_mode.py,
so the phrase is read from config (voice.stop_phrases → $voiceStopPhrase
store, seeded in use-hermes-config) and shown as an info toast when a
voice conversation starts. i18n: en/ja/zh/zh-hant/ar.
Replace one-shot VAD calibration with a rolling deque window that
continuously recalibrates the noise floor throughout TTS playback,
preventing false barge-in triggers from stale calibration. Add a
grace period before VAD activates so TTS playback establishes first.
Suppress duplicate text rendering when token streaming is enabled.
Mirror the barge-in and TTS stream stop logic to the TUI gateway path
so both CLI and gateway use the same VAD semantics.
Rolling-window VAD:
- 90th percentile of rolling window (~3s) for noise floor
- 8x multiplier (was 5x) for TTS volume variation headroom
- 4000 RMS trigger ceiling so genuine speech can still trip
- min_floor clamped to SILENCE_RMS_THRESHOLD * 2
- sustained_ms=1000, calibration_ms=800
Barge-in grace period (barge_in_grace_seconds, default 2.0s):
Delays VAD activation so TTS playback establishes before the mic opens.
Duplicate render suppression:
When streaming_enabled, pass display_callback=None to
stream_tts_to_speaker so the token stream is the sole display path.
TUI gateway mirror:
Mirror _tts_stream_stop and _tts_stream_barge_in_monitor changes to
tui_gateway/server.py so both code paths use the same VAD parameters,
grace period, and TTS CUT diagnostic logging.
Profile-scoped session DB and MoA progress events in tui_gateway/server.py
were necessitated by the TTS pipeline changes affecting session state
and event routing.
Normal-exit flag and TTS CUT diagnostic logging at all cut paths.
Regression tests:
- test_quiet_then_loud_playback_does_not_trip
- test_8x_multiplier_absorbs_tts_volume_spikes
- test_trigger_ceiling_lets_genuine_speech_trip
- test_silence_calibration_does_not_false_trip_on_tts
- test_tts_stream_stop_latches_interruption_for_next_turn
- test_tts_stream_stop_after_natural_finish_does_not_latch
- Profile-scoped session DB tests (10 tests)
Saying OR typing a configured stop phrase (voice.stop_phrases, default
"stop") now ends the voice chat everywhere, not just classic CLI PTT:
- hermes_cli/voice.py: new explicit on_stop_phrase callback through
start_continuous/stop_continuous. The force-transcribe path previously
DISCARDED the stop phrase silently — with auto_restart=False the client
re-arms the next capture, so the conversation never ended. Both halt
paths now fire on_stop_phrase (fallback: on_silent_limit for legacy
callers) as user intent, distinct from the no-speech timeout.
- tui_gateway/server.py: voice.record wires on_stop_phrase and emits
voice.transcript {stop_phrase: true} after flipping HERMES_VOICE(_TTS)
off and stopping streaming TTS — same teardown as /voice off. The TTS
barge-in monitor stop-checks its transcript too. prompt.submit consumes
a TYPED bare stop phrase at the server-side choke point when voice mode
is on (returns {voice_stopped: true}, no turn starts).
- ui-tui: voice.transcript {stop_phrase} ends voice mode with a clear
'voice chat ended' notice (distinct from the no-speech-limit message);
submitPrompt releases the busy latch on a consumed voice_stopped reply.
- cli.py: _typed_voice_stop in process_loop — typing a bare stop phrase
while voice mode/continuous is active ends voice mode instead of
sending 'stop' to the agent; typed 'stop' outside voice mode is
unchanged. Voice transcripts skip the check (already stop-checked).
- desktop: interceptsTypedVoiceStop — the composer's onSubmit ends the
live voice conversation (same path as clicking end on the pill) when a
bare stop command is typed with no attachments; renderer-owned loop, so
handled client-side like the existing spoken isVoiceStopCommand.
- tools/voice_mode.py: transcribe_recording never lets the Whisper
hallucination filter swallow a configured stop phrase (e.g. 'bye'
configured as a stop phrase is both a hallucination-blocklist entry and
a stop phrase — stop-phrase check now wins).
Tests: continuous-loop signal (sabotage-verified), force-transcribe stop
signal + legacy fallback, hallucination-filter ordering, typed-stop CLI
unit tests (voice on/off/longer text), prompt.submit typed-stop gateway
tests, TUI vitest for stop_phrase event handling, desktop vitest for the
typed-stop interceptor.
## Summary
- Spawn system audio players (`ffplay` / `afplay` / `aplay`) with `hermes_subprocess_env(inherit_credentials=False)`.
- Prevent gateway tokens and provider API keys from leaking into OS media helpers.
- Add a regression test asserting scrubbed env on `Popen`.
## Salvage / credit
Sibling of #70342 / incomplete #56332 (TTS/STT command scrub) on the voice-mode playback path.
Two in-house micro-fixes from issue triage:
- #49883: voice.beep_enabled gates in cli.py and hermes_cli/voice.py used
bool() on the config value, so a quoted YAML string like "false" or
"off" kept beeps on. Route through utils.is_truthy_value.
- #18432: AudioRecorder.start() collapsed OSError from _import_audio into
the 'pip install sounddevice numpy' hint — but OSError means the
PortAudio SHARED LIBRARY is missing, which pip cannot fix. Mirror
detect_audio_environment's system-package hint (libportaudio2 /
brew portaudio / Termux pkg install portaudio) on that path.
Fixes#49883Fixes#18432
When Hermes runs inside its bundled venv (~/.hermes/profiles/<name>/
hermes-agent/venv/), the bare 'pip install sounddevice numpy' hint in
_voice_capture_install_hint() told users to install into whichever
Python their shell resolves first — on macOS that is often the system
Python under Rosetta, a completely separate site-packages tree with
incompatible wheel arches.
Detect venv via sys.prefix != sys.base_prefix and return the full path
to the venv's pip binary (Path(sys.prefix)/bin/pip) when available.
Falls back to the bare hint outside a venv (e.g. bare-metal installs).
Also adds 'from pathlib import Path' which was the only new import.
Addresses review on #62601. Applies a single rule — no sounddevice for
audio OUTPUT on macOS (PortAudio/CoreAudio init triggers a
kTCCServiceMediaLibrary prompt) — consistently at all three output sites:
- play_audio_file: WAV playback (already routed to afplay) now uses the
shared _sounddevice_output_allowed() helper.
- play_beep: synthesize the tone with numpy only, then on macOS play it
via a temp WAV through afplay instead of sounddevice.
- stream_tts_to_speaker (tts_tool): on macOS, skip the sounddevice
OutputStream so playback falls through to the existing tempfile/afplay path.
Audio INPUT (recording) is untouched — it legitimately needs mic permission.
Tests: TestMacOSAudioOutputPolicy (voice_mode) proves WAV + beep routing
does not import sounddevice on Darwin and still uses it off Darwin;
test_tts_macos_output proves streaming TTS skips the OutputStream on Darwin.
Existing test_play_wav_via_sounddevice pinned to non-Darwin for determinism.
Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
On macOS, initializing PortAudio/CoreAudio via sounddevice triggers a
kTCCServiceMediaLibrary permission dialog even when no media-library
access is needed. afplay already handles WAV (and every other format)
natively, so route macOS playback straight to it and keep sounddevice
for other platforms.
Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
\`pm list packages com.termux.api\` is the canonical way to detect the
Termux:API Android app, but on some devices it gives a false negative
even when the app is installed and \`termux-microphone-record\` runs
fine — the symptom reported in #31015 (\`/voice on\` complaining
"Termux:API Android app is not installed").
Replace the single probe with a graded strategy:
1. Try \`pm list packages com.termux.api\` (current behaviour).
2. If \`pm\` isn't on PATH or returns non-zero, fall back to
\`cmd package list packages com.termux.api\` — the modern Android
API 28+ equivalent that's present on devices where \`pm\` is gone.
3. If both probes are inconclusive (binary missing, permission
denied, timeout, or non-zero exit) and \`termux-microphone-record\`
is on PATH, trust the binary. The CLI ships in the \`termux-api\`
package which is essentially only useful with the Android app
installed; users who installed it deliberately almost always have
the app too.
Polarity matters: a false negative on this gate blocks \`/voice on\`
entirely (the user-reported symptom), while a false positive only
surfaces a precise runtime error from the binary itself when it
tries to talk to the missing app — strictly more actionable.
The clean-probe-but-no-package case still returns False, so the
existing "Termux:API CLI installed without the app" warning still
fires when the package manager *can* tell us the app is missing.
Refs: NousResearch/hermes-agent#31015
Ports #63768 forward onto current main per teknium1's review.
On WSL2 without a PulseAudio bridge, ffplay and aplay have no audio
device and TTS playback silently fails (issue #17608). When
powershell.exe and ffmpeg are available, convert the audio to a
uniquely-named WAV in the Windows %TEMP% directory and play it via
Media.SoundPlayer.
Per review, this fixes two gaps in the original port:
1. Exit-status masking: the cleanup subshell was
'( ffmpeg && powershell ); rm -f wav' -- the shell's exit status is
the LAST command's (rm -f, which is always 0), so a real
ffmpeg/PowerShell failure could never be detected by the rc-checking
fallback logic added to the player loop. Now captures the real
status before cleanup and re-exits with it:
'( ffmpeg && powershell ); rc=0; rm -f wav; exit '.
2. The no-Pulse WSL gate in detect_audio_environment() still hard-blocked
voice mode entirely (input AND output) even when the PowerShell
fallback made TTS output viable. Added _wsl_powershell_tts_available()
and use it to downgrade the WSL-without-Pulse case from a hard
'warnings' block to a non-blocking 'notices' entry when the fallback
is available -- the same PulseAudio-bridge recording guidance is still
surfaced (mic capture genuinely still needs it), it just no longer
blocks /voice on for TTS-only usage. cli.py's existing
env_check['available'] gate needed no changes since it already
respects this flag.
Also fixed the flaky uniqueness test (the original asserted
len(filenames) >= 2, which passed trivially on zero captured
filenames) and added a real fallback-triggering regression test for
the exit-status fix.
10 new/fixed tests pass in TestWSL2PowerShellFallback and the new
TestWSLAudioEnvironmentGate; 80/80 in the full tests/tools/test_voice_mode.py file.
WSLg RDP audio has two issues causing crackling:
1. systemd-timesyncd clock adjustments jitter PulseAudio timing
(microsoft/wslg#1257) — user action: stop the service
2. Cold-start RDP connection drops first ~100ms of audio packets
before the virtual channel stabilises
Fix (automated, WSL-only):
- Detect WSL via /proc/version 'microsoft' marker
- Prepend 100ms silence + apply 100ms fade-in to audio
- Append 50ms silence tail for clean stream teardown
- Set blocksize=4096 (default auto ~1024 is too small for RDP)
- All in a single continuous sd.play() buffer
Non-WSL paths unchanged.
Closes#38893
AudioRecorder hard-codes SAMPLE_RATE (16 kHz) when opening the input
stream, but some capture devices (e.g. USB microphones exposed through
ALSA hw) reject 16 kHz outright — sd.InputStream fails with
PaErrorCode -9997 (Invalid sample rate) and voice recording is broken.
Query the default input device for its native default_samplerate at
recording start and open the stream / write the WAV at that rate,
falling back to the Whisper-friendly 16 kHz constant when the backend
does not expose a usable rate. STT providers accept standard WAV rates,
so downstream transcription is unaffected.
Closes#55908. The CLI voice-mode beep amplitude is hardcoded at 0.3 inside
tools.voice_mode:play_beep(), which makes the record start/stop cues too
quiet on low-volume systems and headphones. Users couldn't adjust it
without editing source.
Move the literal into a configurable voice.beep_volume setting (clamped to
0.0-1.0, default 0.3 to preserve prior behaviour). The new
_get_beep_volume() helper reads via the same load_config() pattern used by
cli.py's _voice_beeps_enabled() and hermes_cli/voice.py's _beeps_enabled(),
keeps bools / out-of-range / non-numeric / NaN values safely on the default,
and falls back silently if config can't load so the audio cue never breaks
the voice loop on a degenerate config.yaml.
Covered by tests/tools/test_voice_mode.py:
- TestGetBeepVolume (12 cases: missing key, custom value, boundary 0.0/1.0,
out-of-range clamp, type coercion, bool guard, NaN guard, exception
guard, dict-typed voice section)
- TestPlayBeepVolumeWiring (guards against re-introducing a hardcoded 0.3
literal in play_beep)
Docs: website/docs/user-guide/configuration.md mentions the new key.
Other locale translations (zh-Hans etc.) intentionally untouched —
handled by the regular i18n sync pipeline as a separate change.
No change in default behaviour: existing users hear exactly the same beep.
The /voice status command only checked for 'local', 'groq', and 'openai'
providers. Any other valid provider (local_command, mistral, xai,
elevenlabs, or custom command providers) fell through to the generic
MISSING message — even when transcription worked perfectly.
- Import _has_any_command_stt_provider (already defined, never imported)
- Add elif branches for local_command, mistral, xai, elevenlabs
- Add generic catch-all via _has_any_command_stt_provider() for
arbitrary custom command providers
Saying EXACTLY a configured stop phrase (default: 'stop') and nothing
else now ends the voice conversation instead of being sent to the agent
as a prompt. Match is deliberately strict — whole utterance,
case-insensitive, surrounding punctuation stripped — so 'stop doing
that and try X' still reaches the agent.
- tools/voice_mode.py: is_voice_stop_phrase() + voice.stop_phrases
config loader (default ['stop'], [] disables, malformed config falls
back safely).
- hermes_cli/voice.py: shared continuous loop (TUI + desktop) halts on
a stop phrase exactly like the silent-cycle limit (fires
on_silent_limit so every UI turns voice off); stop_continuous
force-transcribe path swallows the phrase without counting a silent
cycle.
- cli.py: classic CLI push-to-talk/continuous path and barge-in
utterance path disable voice mode on a stop phrase.
- Config default voice.stop_phrases: ['stop']; docs updated.
Tests: tests/tools/test_voice_stop_phrase.py (27 tests,
sabotage-verified: loop test fails when detection is disabled);
voice suites green (110 + 44 passed).
On Windows with Chinese locale (GBK), subprocess.run(text=True) without
explicit encoding causes UnicodeDecodeError crashes. This fix adds
encoding='utf-8', errors='replace' to all subprocess.run() and
subprocess.Popen() calls that use text=True across 76 non-test Python files.
Fixes#53428 (master tracker for Windows GBK locale crash).
Note: credential_pool.py and electron changes excluded per reviewer request —
those will be submitted as separate focused PRs.
listen_for_speech(): sustained-RMS speech detection with the noise
floor calibrated against audible playback (speaker bleed doesn't
self-trigger). With capture=True it keeps a rolling pre-roll buffer and
records through to a silence endpoint, so the interruption is
transcribed from its first syllable — detection alone loses the opening
words to the detector's sustain window plus the mic re-open.
transcribe_recording() maps no_speech provider results to a successful
empty transcript (silence, not an error).
Removes Homebrew and PyPI wheel/sdist as Hermes distribution paths while
preserving the supported source, Docker, and Nix workflows.
Changes:
- Removes the Homebrew formula, PyPI publish workflow, sdist manifest
(MANIFEST.in), and wheel/sdist release-attachment logic from scripts/release.py.
- Keeps setuptools metadata and entry points required by editable installs
and Docker/Nix builds, but adds a setup.py guard that rejects wheel/sdist
builds outside a sealed Nix derivation (HERMES_NIX_BUILD=1).
- Removes pip/Homebrew install detection, PyPI update checks, the pip
self-update path, the deprecation-banner state, the postinstall subcommand,
wheel data-directory fallbacks in agent/i18n.py and hermes_constants.py,
and the ACP Registry manifest/version-lockstep release logic.
- Adds /nix/store/ path detection so `nix run` / `nix profile install`
installs (which don't set HERMES_MANAGED) are correctly identified as
"nix" rather than falling through to "git"/"unknown".
- Retired install-method values ("pip", "homebrew") in existing
.install_method stamps (both code-scoped and home-scoped) are ignored by
the allowlist reader and fall through to "unknown" instead of resurrecting
a retired enum value.
- Updates Nix packaging to ship bare runtime data (locales, optional-mcps)
through store symlinks and wrapper env vars instead of wheel data-files.
- Removes the ACP Registry manifest/icon and their version-lockstep tests.
- Deletes or rewrites packaging, pip-update, Homebrew, and ACP Registry
tests; adds parametrized coverage for the packaging build guard covering
BOTH sdist and wheel paths (the guards live in separate cmdclass entries
— a passing sdist test proves nothing about the wheel path).
- Updates installation/platform documentation and related user-facing copy.
- Adjusts the supply-chain scan so deleted install-hook files do not trigger
a finding, while additions or modifications still require the existing
ci-reviewed label gate.
Supported installation paths (unchanged):
- git installer (install.sh)
- Docker
- Nix/NixOS
- editable development installs (uv sync, uv pip install -e ., pip install -e .)
When Hermes runs in TUI mode, the gateway child process communicates with
the Node.js parent over a JSON-RPC protocol on stdin. Subprocess calls that
inherit this stdin fd can trigger a race condition where the child's stdin
read returns EOF, causing the gateway to exit cleanly (exit code 0) mid-tool-
execution.
This is the same root cause as issue #14036 (byterover plugin) and PR #39257
(SSH environment backend). This commit applies the fix — stdin=subprocess.DEVNULL
— to all 85 subprocess.run() and subprocess.Popen() calls that execute inside
the TUI gateway child process.
Scope: TUI-context code only (agent/, tools/, plugins/, tui_gateway/server.py).
CLI code (cli.py, hermes_cli/), tests, scripts, and gateway process management
are excluded — they don't run inside the TUI child and inherit the terminal's
stdin, not the JSON-RPC pipe.
85 call sites across 28 files. All files pass syntax check.
SSH sessions hard-failed voice mode on the presence of SSH_* env vars
alone, even when a PulseAudio/PipeWire server is running on the host and
audio works (ffplay/aplay/pw-play -> pulseaudio). Probe the default
sound-server sockets (PULSE_SERVER unix path, PULSE_RUNTIME_PATH/native,
$XDG_RUNTIME_DIR/{pulse/native,pipewire-0}) and actually connect() so a
stale socket doesn't count; downgrade the SSH branch to a notice when
audio is reachable. Mirrors the existing Docker/WSL forwarding handling.
Fixes#35622
* docs(voice): use `uv pip install faster-whisper` in STT install hints
Three runtime messages told users to `pip install faster-whisper`
(reported in #29782 for the gateway STT failure message under
Telegram-in-Docker, where the user hit `bash: pip: command not
found`). The Hermes Docker image is built on `ghcr.io/astral-sh/uv`
with a uv-managed venv that doesn't ship `pip` on PATH; users on
modern `uv tool install` / `uv venv` installs see the same problem.
The canonical install command in this repo is `uv pip install`
(see `tools/lazy_deps.py:509` `feature_install_command()`), which
works in Docker (uv image), in `uv tool install` venvs, and in
pip-based venvs that already have uv on PATH.
Changed three locations to match:
- `gateway/run.py` — Telegram/Discord/Slack/WhatsApp/etc. voice
reply when no STT provider is configured. Suggests
`uv pip install faster-whisper` and notes that
`pip install faster-whisper` also works if `pip` is on PATH.
- `tools/voice_mode.py` — `/voice` status line for missing STT.
- `cli.py` — Voice-mode startup error, "Option 1".
No behavior change beyond the user-facing text. No production
code path was touched.
* docs(voice): add pip fallback to cli + voice_mode STT hints
Copilot flagged that cli.py and tools/voice_mode.py recommend
`uv pip install faster-whisper` without a fallback for environments
where uv isn't on PATH. The gateway/run.py message already lists
`pip install faster-whisper` as an alternative; this commit aligns
the two remaining call sites to match.
Addresses inline Copilot review on #29800.
---------
Co-authored-by: briandevans <252620095+briandevans@users.noreply.github.com>
In WSL2, sounddevice.query_devices() returns [] even when the
PulseAudio bridge is functional. The existing code already handled
the case where the query itself raises an exception, but it missed
the empty-list case.
This change treats an empty device list as non-fatal in WSL when
PULSE_SERVER is configured, matching the existing exception-handler
behavior.
Fixes: WSL users seeing 'No audio input/output devices detected'
even though paplay/arecord work fine.
detect_audio_environment() unconditionally added a hard warning when
running inside a container, blocking /voice on even when the host audio
socket was correctly forwarded (PulseAudio or PipeWire) and sounddevice
could enumerate devices.
Mirror the existing WSL/PulseAudio handling: if PULSE_SERVER or
PIPEWIRE_REMOTE is set, downgrade to a notice and let the audio backend
decide. When neither is set, keep the block but extend the message with
the exact -v / -e flags users need.
Closes#21203
Closes the last Python-on-Windows UTF-8 exposure by making every
text-mode open() call explicit about its encoding.
Before: on Windows, bare open(path, 'r') defaults to the system
locale encoding (cp1252 on US-locale installs). That means reading
any config/yaml/markdown/json file with non-ASCII content either
crashes with UnicodeDecodeError or silently mis-decodes bytes.
After: all 89 affected call sites in production code now pass
encoding='utf-8' explicitly. Works identically on every platform
and every locale, no surprise behavior.
Mechanical sweep via:
ruff check --preview --extend-select PLW1514 --unsafe-fixes --fix --exclude 'tests,venv,.venv,node_modules,website,optional-skills, skills,tinker-atropos,plugins' .
All 89 fixes have the same shape: open(x) or open(x, mode) became
open(x, encoding='utf-8') or open(x, mode, encoding='utf-8'). Nothing
else changed. Every modified file still parses and the Windows/sandbox
test suite is still green (85 passed, 14 skipped, 0 failed across
tests/tools/test_code_execution_windows_env.py +
tests/tools/test_code_execution_modes.py + tests/tools/test_env_passthrough.py +
tests/test_hermes_bootstrap.py).
Scope notes:
- tests/ excluded: test fixtures can use locale encoding intentionally
(exercising edge cases). If we want to tighten tests later that's
a separate PR.
- plugins/ excluded: plugin-specific conventions may differ; plugin
authors own their code.
- optional-skills/ and skills/ excluded: skill scripts are user-authored
and we don't want to mass-edit them.
- website/ and tinker-atropos/ excluded: vendored / generated content.
46 files touched, 89 +/- lines (symmetric replacement). No behavior
change on POSIX or on Windows when the file is ASCII; bug fix on
Windows when the file contains non-ASCII.
Error messages that tell users to install optional extras now use
{sys.executable} -m pip install ... instead of a bare 'pip install
hermes-agent[extra]' string. Under the curl installer, bare 'pip'
resolves to system pip, which either fails with PEP 668
externally-managed-environment or installs into the wrong Python.
Affects: hermes dashboard, hermes web server startup, mcp_serve,
hermes doctor Bedrock check, CLI voice mode, voice_mode tool runtime
error, Discord voice-channel join failure message.
Centralize container detection in hermes_constants.is_container() with
process-lifetime caching, matching existing is_wsl()/is_termux() patterns.
Dedup _is_inside_container() in config.py to delegate to the new function.
Add _run_systemctl() wrapper that converts FileNotFoundError to RuntimeError
for defense-in-depth — all 10 bare subprocess.run(_systemctl_cmd(...)) call
sites now route through it.
Make supports_systemd_services() return False in containers and when
systemctl binary is absent (shutil.which check).
Add Docker-specific guidance in gateway_command() for install/uninstall/start
subcommands — exit 0 with helpful instructions instead of crashing.
Make 'hermes status' show 'Manager: docker (foreground)' and 'hermes dump'
show 'running (docker, pid N)' inside containers.
Fix setup_gateway() to use supports_systemd instead of _is_linux for all
systemd-related branches, and show Docker restart policy instructions in
containers.
Replace inline /.dockerenv check in voice_mode.py with is_container().
Fixes#7420
Co-authored-by: teknium1 <teknium1@users.noreply.github.com>
Automated dead code audit using vulture + coverage.py + ast-grep intersection,
confirmed by Opus deep verification pass. Every symbol verified to have zero
production callers (test imports excluded from reachability analysis).
Removes ~1,534 lines of dead production code across 46 files and ~1,382 lines
of stale test code. 3 entire files deleted (agent/builtin_memory_provider.py,
hermes_cli/checklist.py, tests/hermes_cli/test_setup_model_selection.py).
Co-authored-by: alt-glitch <balyan.sid@gmail.com>
Replace 6 identical copies of the Termux detection function across
cli.py, browser_tool.py, voice_mode.py, status.py, doctor.py, and
gateway.py with a single shared implementation in hermes_constants.py.
Each call site imports with its original local name to preserve all
existing callers (internal references and test monkeypatches).
kshitij
chelsealong
Teknium
beardedeagle
峯岸 亮
RedClaus
simonmmafs
Simon
xxxigm
ygd58
Ivan Kharitonov
Slippy87
Brice
Kewe63
Zehua Wang
Charles Cha
Stoltemberg
brooklyn!
ethernet
m4dni5
Dusk
Brian D. Evans
Ben Barclay
helix4u
RhombusMaximus
kshitij
Wesley Simplicio
alt-glitch
alt-glitch
Teknium
adybag14-cyber