Second, deeper pass over tools/gateway/hermes_cli plus first pass over
the trees wave 1 missed (acp, acp_adapter, skills, computer_use, docker,
dashboard, conformance, monitoring, secret_sources, hermes_state,
providers). Same rubric as wave 1 (AGENTS.md test policy); security,
alternation/caching invariants, issue-number regressions, and E2E kept.
Real test-quality fixes found and rooted out along the way:
- tests/tools/test_command_guards.py made real auxiliary-LLM HTTPS calls
(DEFAULT_CONFIG smart-approval leaked in) — pinned approval
mode=manual via autouse fixture: 17.4s → 0.4s.
- test_model_switch_custom_providers.py / test_user_providers_model_switch.py
silently probed live provider catalogs (~2s/test) — stubbed
cached_provider_model_ids/provider_model_ids/fetch_api_models.
- test_telegram_noise_filter.py: 15-platform copy-paste matrix over
shared gateway.run logic → 3 representative platforms (55s → 3.9s).
- test_gateway_shutdown.py: stop()'s 5s interrupt-deadline loop spun on
MagicMock agents — interrupt.side_effect now clears _running_agents
(22s → 1.0s).
- test_gateway_inactivity_timeout.py poll-harness timings shrunk 3-5x
(24s → 1.1s); test_mcp_stability.py backoff/SIGTERM-grace sleeps
patched (15.4s → 2.5s); test_async_delegation.py negative-drain wait
5s → 0.5s.
- test_telegram_init_deadline.py: loop-block margin restored to 1.0s
with rationale comment — the watchdog-dump assertion needs the loop
blocked well past deadline+grace under parallel load (flaked once in
the 40-worker verification run at a 0.2s margin).
Verification: full hermetic suite via scripts/run_tests.sh —
2,438 files, 21,718 tests passed, 0 failed, 293.9s wall.
Suite totals vs original baseline: 46,820 → 19,757 test functions
(−57.8%), wall 583.5s → 293.9s (−50%), subprocess CPU 13,564s → 11,623s.
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.
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)
Review follow-up: test_micro_pause_tolerance_during_speech was left on real
sleeps, so it kept the wall-clock dependency the rest of this PR removes.
Its three sleeps (50 ms, 50 ms, 60 ms) feed the same monotonic gate at
tools/voice_mode.py:561 that the other two tests were converted for.
It was measured rather than assumed before being left out: 0 failures in 40
runs, because on Windows sleep() rounds up to the timer tick, so sleep(0.05)
really costs ~62.5 ms and the three sleeps clear the 150 ms gate by ~37.5 ms
-- wider than the 15.625 ms GetTickCount64() quantum that made the other two
flake. That margin is an accident of sleep granularity rather than anything
the test guarantees, so it is not worth keeping the bet.
Driving it from fake_clock makes the timeline exact (0.05 + 0.05 + 0.06 =
0.16 against the 0.15 gate, dip 0.05 under the 0.1 tolerance) and drops the
real sleeping from the suite. Mutation-checked: widening the gate at :561
fails the test.
test_silence_callback_fires_after_speech_then_silence and
test_custom_threshold_and_duration space their audio frames with
time.sleep(0.06) and check the result against 50 ms thresholds. That
leaves a 10 ms margin, which only survives if the clock is finer-grained
than the margin — and on Windows it isn't. time.monotonic() there is
GetTickCount64() with 15.625 ms resolution until CPython 3.13 moved it to
QueryPerformanceCounter(), so a sleep that really lasts 62.5 ms measures
as 46.9 ms often enough to matter, landing under the threshold. Depending
on which sleep got clipped, either the speech-confirm gate misses or the
silence timer never matures — which is why the same flake shows up on two
different asserts.
Both tests now advance a hand-driven clock instead of sleeping, so the
arithmetic is exact on every platform and neither test waits on real time.
The fixture patches the `time` name inside tools.voice_mode rather than
time.monotonic itself: voice_mode.time IS the stdlib module, so patching
through it would swap the clock out from under every other importer for
the duration of the test.
Measured on Windows 11 / Python 3.11.9, 20 runs of each test: 6/20 and
3/20 failed before, 0/30 after. The same machine on Python 3.13, where
monotonic() is QueryPerformanceCounter(), never failed either test — that
comparison is what pinned the clock resolution as the cause rather than
the detection logic. Linux CI never sees this: clock_gettime is
nanosecond-resolution there.
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
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>
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.
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.
Review follow-up: the sweeper is right that the first commit only cured
half the dead config — the TUI gateway builds its recorder params
explicitly, so the cap never reached recordings started from the TUI.
- start_continuous() grows a max_recording_seconds param (default 0.0 =
disabled, so existing callers keep today's behaviour) and applies it to
the shared recorder next to the silence params
- tui_gateway voice.record start forwards the validated cap; corruption
semantics now mirror the silence params everywhere: non-numeric/bool
falls back to the documented 120 default (a hand-edited
`max_recording_seconds: true` must not become a 1-second cap), while
an explicit numeric <= 0 disables the cap
- cli.py wiring updated to the same corruption semantics
New coverage, per review:
- TestMaxRecordingCap drives the mocked InputStream callback past the
cap during continuous loud speech (the silence branch physically can't
fire there) and asserts the one-shot callback fires exactly once; plus
the disabled-cap negative
- TestMaxRecordingSecondsConfigReal pins the CLI config assignment for
the valid / zero / bool / garbage cases via _voice_start_recording
- test_voice_record_start_forwards_max_recording_seconds pins the TUI
forwarding for the same matrix
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).
* feat(attribution): conflict-free contributor mappings via contributors/emails/ directory
The AUTHOR_MAP dict in scripts/release.py was a merge-conflict magnet:
every concurrent salvage PR appended entries to the same lines of the
same file, so parallel PRs re-conflicted on every merge to main.
New system: one file per email under contributors/emails/ — filename is
the commit-author email, first non-comment line is the GitHub login.
File additions never conflict, so any number of PRs can add mappings
concurrently.
- scripts/release.py: AUTHOR_MAP is now LEGACY_AUTHOR_MAP (frozen)
merged with the directory at import time (directory wins). All
existing consumers (resolve_author, contributor_audit.py) unchanged.
- scripts/add_contributor.py: idempotent CLI to add a mapping; refuses
conflicting reassignments (incl. against the legacy map), validates
email/login shapes.
- contributor-check.yml: attribution gate now accepts a mapping file OR
a legacy entry; failure message prints the exact add_contributor
command. Also auto-resolves bare <login>@users.noreply.github.com
emails is intentionally NOT added (kept id+login form only, matching
previous behavior).
- contributor_audit.py: guidance now points at add_contributor.py.
- tests/scripts/test_contributor_map.py: 12 tests covering loader,
merge precedence, CLI idempotency/conflict/validation, subprocess E2E.
* feat(ci): one-shot per-file flake retry in the parallel test runner
A failing test FILE is re-run once in a fresh subprocess. Pass-on-retry
counts as green but is loudly reported in a '⚠ FLAKY' summary section
(with both attempts' output preserved) so the flake gets fixed instead
of eating a full-run rerun. Deterministic failures fail both attempts —
regressions cannot be laundered green.
- --file-retries N / HERMES_TEST_FILE_RETRIES (default 1, 0 disables)
- E2E verified: simulated first-run-fail flake goes green with banner;
deterministic failure still exits 1; retries=0 restores old behavior.
This converts the dominant CI failure mode (one timing-sensitive test
flaking a 4600-test shard, requiring a manual 10-minute rerun and an
agent triage loop) into a self-healing retry that costs one file's
runtime.
* test(approval): loosen wall-clock perf bounds 0.15s -> 2.0s
These guard against catastrophic regex backtracking (seconds-to-minutes
class), but 0.15s is within scheduler-stall noise on loaded shared CI
runners — test_max_accepted_separator_free_input_is_fast failed a CI
shard this week on runner load alone. 2.0s still catches the regression
class with zero flake surface.
* fix(ci): job timeouts everywhere + retries on all network installs
Reliability pass over every workflow:
- timeout-minutes on all 21 jobs that lacked one (a hung job previously
burned the 6-hour default runner budget)
- ./.github/actions/retry wrapped around every network-fetching install
that lacked it: pip installs (deploy-site, skills-index), npm ci
(deploy-site website, upload_to_pypi web + ui-tui), uv sync (docker
test deps). Deterministic build steps (npm run build) deliberately
NOT retried — split into separate steps so a real build failure fails
fast instead of retrying 3x.
* docs(agents): document the file-retry flake policy
* fix(ci): curl retries on deploy hook + skills-index probe
* fix(ci): kill the remaining transient-failure classes in workflows + Dockerfile
From the workflow reliability audit:
- tests.yml: duration-cache restore had NO restore-keys while saves use
run_id-suffixed keys — the cache never matched once, so LPT slicing
always ran blind and unbalanced slices pushed heavy files toward the
per-file timeout. One-line restore-keys fixes slice balancing.
- Label gates (lint ci-reviewed, supply-chain mcp-catalog-reviewed):
'gh pr view || true' turned an API blip into 'label absent' → false
BLOCKING failure. Now 3x retry, and API failure is reported as an API
failure instead of a missing label.
- detect-changes action: compare API retried before failing open (was
silently running all lanes on any blip).
- uv-lockfile-check: 'uv lock --check' resolves against PyPI — retried
so registry blips don't read as 'lockfile stale'.
- docker.yml merge job: imagetools create retried (Docker Hub eventual
consistency on just-pushed digests).
- Dockerfile: apt-get Acquire::Retries=3; s6-overlay ADDs converted to
curl --retry 3 (ADD cannot retry; checksums still enforced); npm
--fetch-retries=5; playwright chromium fetch retried 3x.
- Advisory artifact uploads (per-slice durations, ci-timings report)
get continue-on-error so an artifact-service blip can't fail a green
test slice.
* fix(tests): kill the two root-cause flakes — leaking pre-warm timer + env-dependent provider list
- test_tui_gateway_server.py: session.create / non-eager session.resume
arm a 50ms threading.Timer (_schedule_agent_build) that outlives its
test and fires into the NEXT test's _make_agent mock, racily
corrupting captured state (the recurring session_resume shard
failures). Replaced the per-test whack-a-mole stub with a module-wide
autouse fixture; the 3 worker-lifecycle tests that genuinely need the
deferred build opt back in via @pytest.mark.real_agent_prewarm (new
marker in pyproject).
- test_api_key_providers.py: PROVIDER_ENV_VARS is now derived from the
live PROVIDER_REGISTRY instead of a hand-list that had drifted
(missing HF_TOKEN / DEEPINFRA_API_KEY) — resolve_provider('auto')
tests failed on any machine with HF_TOKEN exported. E2E-verified with
HF_TOKEN/DEEPINFRA_API_KEY set: 42/42 pass.
* test: de-flake 30 timing-sensitive test files for loaded CI runners
Root-cause fixes from the flake audit (session-DB mining + repo sweep):
Event-based sync instead of sleep-sync:
- title_generator: mock sets threading.Event, wait(10) replaces
sleep(0.3) hoping the daemon thread got scheduled
- docker zombie_reaping / profile_gateway: poll-for-state helpers
replace fixed 1-3s sleeps (s6 transitions + SIGCHLD reaping are async)
- process_registry tree test: select()-bounded readline replaces an
unbounded blocking read (parent wedge now fails THIS test with a clear
message instead of an opaque rc=124 file kill); SIGTERM grace 1s->2s
(the 1s partition window mid-interpreter-startup is how a child PID
escaped the live-system guard in CI)
Timeout raises (loaded 8-way-sliced runners see ~5s scheduling floors;
all of these complete in ms-to-1s when healthy so the raises cost
nothing on green runs):
- subprocess/thread waits <= 2s raised to 10-15s across mcp_tool,
mcp_circuit_breaker, mcp_reconnect_retry_reset, mcp_parked_self_probe,
mcp_cancelled_error_propagation, registry, clarify_gateway, interrupt,
voice_cli_integration, docker_environment, session_store_lock_io,
planned_stop_watcher, cli_interrupt_subagent, thread_scoped_output
(joins now also assert not is_alive() so stragglers fail loudly)
- wall-clock discrimination ceilings loosened where the guarded hang is
10x larger: local_background_child_hang 4s->10s, interrupt_cleanup
setup 5s->20s + pgid-exit 30s->60s, mcp_stability grandchild spinup
5s->15s, protocol/gil-starvation fast-handler 0.5s->2s,
iso_certify_seam 1.5s->5s, wait_for_mcp_discovery 0.1s->1s
- narrow assertion windows widened: honcho first-turn wait 0.4..0.65 ->
0.25..2.0 (property is bounded-not-hung, not an exact wall-clock);
compression fork-lock TTL 1s->3s (12 refresh chances per lease);
compression-lock expiry margins symmetric (ttl 0.05->0.5, sleep 1.0)
- telegram hung-DNS bound 1.0->1.4 (fake hang is 1.5s — must stay under)
* fix(tests): repair indentation from de-flake batch edit
* fix(tests): harden env isolation and replace remaining sleep-sync races
The full 42k-test run and complete npm check surfaced three more classes:
- Environment isolation: local ~/.honcho defaultHost and SSH_* variables
leaked into Python/TUI tests. Pin the default Honcho host in the
hermetic fixture, isolate the one fallback test from ~/.honcho, and
blank SSH_* around terminalSetup tests. This flipped 20 false failures
back to deterministic behavior on developer machines.
- Background-thread sleep-sync: Honcho async writer tests patched
time.sleep globally, then busy-polled with that same mocked sleep. Under
full-suite load the poller could starve the writer. Each test now waits
on an Event emitted by the exact flush/retry transition; 30/30 passed
under 15-way contention.
- Desktop streaming: the test slept 80ms and assumed a 500ms timer could
not fire before its assertion. A loaded runner descheduled the test for
>500ms and both chunks arrived. Producer controls now gate second-chunk
and completion transitions explicitly.
Also make file-retry observability complete: a self-healed flaky file now
prints BOTH attempts' full output in the FLAKY summary. Two behavioral
runner tests prove pass-on-retry is green+loud+traceback-preserving, while
a deterministic failure remains red.
* refactor(ci): use gh bot pat, better retries
refactor(ci): use retry action for PR label fetch
the retry action now captures stdout as a step output, so it can serve
double duty: retry + output capture for commands like 'gh pr view' whose
result must be consumed by later steps.
Retry action gains:
- 'stdout' output (heredoc-delimited to preserve newlines)
- tee to temp file so stdout still streams to the job log
- step id 'retry' for output reference
Both lint.yml and supply-chain-audit.yml now use the retry action
directly with 'command: gh pr view ...' and read
steps.<id>.outputs.stdout.
ci: use AUTOFIX_BOT_PAT for all gh CLI / GitHub API auth
Replace secrets.GITHUB_TOKEN and github.token with
secrets.AUTOFIX_BOT_PAT across all workflows and composite actions
that use the gh CLI or GitHub API. The PAT has consistent permissions
across fork PRs (where GITHUB_TOKEN is read-only), avoids API rate
limit sharing with the default token, and is already used by
js-autofix.yml for the same reasons.
19 sites swapped across 9 files:
- lint.yml (3): label fetch, comment post/edit, comment update
- supply-chain-audit.yml (5): scan, critical comment, unbounded dep
comment, label fetch, mcp-catalog comment
- lockfile-diff.yml (1): PR comment post/update
- skills-index-freshness.yml (1): issue creation on degraded probe
- skills-index.yml (2): index build, trigger deploy workflow
- upload_to_pypi.yml (2): release view poll, release upload
- ci.yml (1): timings report
- deploy-site.yml (2): skills index crawl
- detect-changes/action.yml (1): compare API call
---------
Co-authored-by: ethernet <arilotter@gmail.com>
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
Remove unused imports (F401) and duplicate/shadowed import
redefinitions (F811) across the codebase using ruff's safe
autofixes. No behavioral changes -- imports only.
- ~1400 safe autofixes applied across 644 files (net -1072 lines)
- __init__.py re-exports preserved (excluded from F401 removal so
public re-export surfaces stay intact)
- Re-exports that are imported or monkeypatched by tests but look
unused in their defining module are kept with explicit # noqa:
F401 (gateway/run.py load_dotenv; run_agent re-exports from
agent.message_sanitization, agent.context_compressor,
agent.retry_utils, agent.prompt_builder, agent.process_bootstrap,
agent.codex_responses_adapter)
- Unsafe F841 (unused-variable) fixes deliberately skipped -- those
can change behavior when the RHS has side effects
- ruff lints remain disabled in pyproject.toml (only PLW1514 is
selected); this is a one-time cleanup, not a config change
Verification:
- python -m compileall: clean
- pytest --collect-only: all 27161 tests collect (zero import errors)
- core entry points import clean (run_agent, model_tools, cli,
toolsets, hermes_state, batch_runner, gateway)
- static scan: every name any test imports directly from an edited
module still resolves
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
WSL detection was treated as a hard fail, blocking voice mode even when
audio worked via PulseAudio bridge. Now PULSE_SERVER env var presence
makes WSL a soft notice instead of a blocking warning. Device query
failures in WSL with PULSE_SERVER are also treated as non-blocking.
1. Anthropic + ElevenLabs TTS silence: forward full response to TTS
callback for non-streaming providers (choices first, then native
content blocks fallback).
2. Subprocess timeout kill: play_audio_file now kills the process on
TimeoutExpired instead of leaving zombie processes.
3. Discord disconnect cleanup: leave all voice channels before closing
the client to prevent leaked state.
4. Audio stream leak: close InputStream if stream.start() fails.
5. Race condition: read/write _on_silence_stop under lock in audio
callback thread.
6. _vprint force=True: show API error, retry, and truncation messages
even during streaming TTS.
7. _refresh_level lock: read _voice_recording under _voice_lock.
Remove web UI gateway (web.py, tests, docs, toolset, env vars, Platform.WEB
enum) per maintainer request — Nous is building their own official chat UI.
Fix 1: Replace sd.wait() with polling pattern in play_audio_file() to prevent
indefinite hang when audio device stalls (consistent with play_beep()).
Fix 2: Use importlib.util.find_spec() for faster_whisper/openai availability
checks instead of module-level imports that trigger heavy native library
loading (CUDA/cuDNN) at import time.
Fix 3: Remove inspect.signature() hack in _send_voice_reply() — add **kwargs
to Telegram send_voice() so all adapters accept metadata uniformly.
Fix 4: Make session loading resilient to removed platform enum values — skip
entries with unknown platforms instead of crashing the entire gateway.
Voice status was hardcoded to check API keys only. Now uses the actual
provider resolution (local/groq/openai) so it correctly shows
"local faster-whisper" when installed instead of "Groq" or "MISSING".
- Keep InputStream alive across recordings to avoid CoreAudio hang on
repeated open/close cycles on macOS. New _ensure_stream() creates the
stream once; start()/stop()/cancel() only toggle frame collection.
- Add _close_stream_with_timeout() with daemon thread to prevent
stream.stop()/close() from blocking indefinitely.
- Add generation counter to detect stale stream-open completions after
cancel or restart.
- Run recorder.cancel() in background thread from Ctrl+C handler to
keep the event loop responsive.
- Add shutdown() method called on /voice off to release audio resources.
- Fix silence timer reset during active speech: use dip tolerance for
_resume_start tracker so natural speech pauses (< 0.3s) don't prevent
the silence timer from being reset.
- Update tests to match persistent stream behavior.
1. Fully lazy imports: sounddevice, numpy, elevenlabs, edge_tts, and
openai are never imported at module level. Each is imported only when
the feature is explicitly activated, preventing crashes in headless
environments (SSH, Docker, WSL, no PortAudio).
2. No core agent loop changes: streaming TTS path extracted from
_interruptible_api_call() into separate _streaming_api_call() method.
The original method is restored to its upstream form.
3. Configurable key binding: push-to-talk key changed from Ctrl+R
(conflicts with readline reverse-search) to Ctrl+B by default.
Configurable via voice.push_to_talk_key in config.yaml.
4. Environment detection: new detect_audio_environment() function checks
for SSH, Docker, WSL, and missing audio devices before enabling voice
mode. Auto-disables with clear warnings in incompatible environments.
5. Graceful degradation: every audio touchpoint (sd.play, sd.InputStream,
sd.OutputStream) wrapped in try/except with ImportError/OSError
handling. Failures produce warnings, not crashes.
- Audio cues: beep on record start (880Hz), double beep on stop (660Hz)
- Silence detection: auto-stop recording after 3s of silence (RMS-based)
- Continuous mode: auto-restart recording after agent responds
- Ctrl+R starts continuous mode, Ctrl+R during recording exits it
- Waits for TTS to finish before restarting to avoid recording speaker
- Tests: 7 new tests for beep generation and silence detection
The test was failing because GROQ_API_KEY leaked from the environment.
Now both VOICE_TOOLS_OPENAI_KEY and GROQ_API_KEY are removed to
properly test the "no STT key" scenario.
