Two follow-ups from the voice PR (#70509).
1. macOS ARM64 onnx migration. Existing users who pinned
openwakeword.inference_framework=onnx before the tflite fix landed kept a
wake word that arms but never fires (ONNX's embedding model is broken on
Apple Silicon, upstream #336). New resolve_inference_framework() honors an
explicit framework everywhere ONNX actually works, but coerces the one
provably-dead combination (explicit onnx + macOS ARM64) to tflite with a
one-time warning. No config mutation; empty still falls back to the platform
default. Both read sites (engine init + requirements check) route through
the shared resolver.
2. Voice turn-timeout leak. Each listen cycle reassigned turnTimeoutRef
without clearing the prior 60s timer, so a stale timer from an earlier
cycle could fire handleTurn() mid-way through a later listen — after enough
idle re-listens this wedged the loop into a non-re-arming state (the
'voice chat deactivates after ~a minute' report). Clear before re-arm.
Tests: 64 wake tests (added onnx-coercion / intel-kept / tflite-kept /
empty-default cases; updated the stale 'explicit onnx kept on ARM64' test
that encoded the old broken behavior), 39 desktop voice/wake vitest, tsc +
eslint clean.
'photon:any;-;+1555...' targets matched no parser pattern, so
_handle_send bounced them off the channel directory and failed
resolution even though the adapter accepts the GUID verbatim (the
react handler already passed them through). Recognize the DM chat
GUID shape (mirrors the adapter's _DM_CHAT_GUID_RE) in
_parse_target_ref for photon only.
Command providers legitimately reference their own API keys in shell
templates (curl one-liners). The #70342 scrub removes ALL provider keys,
which would break such setups. Add a per-provider env_passthrough list
(TTS + STT) that copies named variables back from the parent env, plus
docs and tests. Scrub stays the default; passthrough is explicit opt-in.
Port the progress-based idle-timeout pattern from _run_command_tts
(PR #50087, @CleanDev-Fix) to _run_command_stt: the timeout resets on
any stdout/stderr output, so a slow-but-alive STT provider survives
while a silently stalled one is killed. Stuck detection stays
progress-based, never wall-clock.
`transcribe_audio` reads a local file and hands it to the configured STT
provider — for the hosted providers (Groq, OpenAI, Mistral, xAI, ElevenLabs)
that ships the file's bytes to a third-party API. The same local-input read
guard was added to image-gen (587be5b5b) and xAI video-gen (104232979) to keep
the agent from feeding credential/secret stores to a provider, but STT was
missed.
Call `get_read_block_error(file_path)` at the top of `transcribe_audio`, before
validation/dispatch, so a `.env`, `auth.json`, `.anthropic_oauth.json`,
`mcp-tokens/`, etc. is refused up front instead of being transcribed (and, for
hosted providers, exfiltrated). This is defense-in-depth, not a security
boundary — the guard's own message says so — but it restores parity with the
image/video-gen tools.
Regression test: a `.env` file is refused with the shared read-guard message
before any provider dispatch (mutation-verified).
HERMES_LOCAL_STT_COMMAND rendered quoted placeholders into a
user-configured template and passed the result to shell=True. Shell
metacharacters in the template therefore remained executable syntax even
though the placeholder values themselves were quoted.
Tokenize the rendered template and invoke it as an argv list while
preserving the existing timeout, closed stdin, and Windows creation flags.
Lock the invocation contract with metacharacter regression coverage and
document explicit shell wrapping for trusted templates that need it.
Salvages #32694
Co-authored-by: Ernest Hysa <takis312@hotmail.com>
Command-type TTS providers validated output_format against a hardcoded
{mp3,wav,ogg,flac} set; any other value was silently coerced back to mp3,
which then mismatched the output path the post-run check expects. This
blocked common ffmpeg-producible containers/codecs — notably m4a (AAC),
the portable choice for WeChat/iOS/mobile voice files — with no
config-only path (only a local source patch, lost on every update).
Widen COMMAND_TTS_OUTPUT_FORMATS to add m4a, aac, amr, opus. This only
permits a command provider to declare these; the user's command still
produces the file (e.g. via ffmpeg). No built-in provider behavior
changes and no new required config.
Update the two tests that pinned the old set, and add a positive case
covering the new formats. Document the supported output_format values.
## 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.
Salvage incomplete #56332: route command TTS/STT through hermes_subprocess_env
while preserving delegated-child lineage, and close the sibling local-whisper
subprocess.run path that still inherited the full process environment.
Co-authored-by: Cursor <cursoragent@cursor.com>
CI (Linux) failed with ModuleNotFoundError: portalocker — it's a
win32-only dependency (concurrent-log-handler chain). Tests now lock
handles via a platform helper (fcntl on POSIX, portalocker on Windows),
mirroring production _try_acquire_mcp_discovery_lock.
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.
Running the suite could take over the machine it ran on. Two real
effects, both now closed:
- **The suite spoke through the speakers.** Once any test drove the
`voice.toggle` RPC with `action="tts"`, the handler set
`HERMES_VOICE_TTS=1` in the *live process environment*, and the flag
outlived that test. Every later test that drove a turn to completion
then fed its final response text to `hermes_cli.voice.speak_text` on a
background thread - real synthesis, real playback, no API key needed
(the default `edge` provider is keyless). A developer heard the fixture
string "partial answer complete" out loud. Because the flag is set from
inside the process, `scripts/run_tests.sh`'s `env -i` never protected
against this.
`tests/conftest.py` now blanks `HERMES_VOICE`/`HERMES_VOICE_TTS` per
test, and a new autouse `_audio_playback_guard` stubs `speak_text` and
its playback binding outright, so the speakers stay shut even inside the
test that sets the flag itself. `@pytest.mark.real_audio_playback` opts
out.
- **The suite launched Chrome.** `tests/tools/test_browser_supervisor.py`
spawned a real browser on any machine with Chrome on `PATH`. Its
docstring promised a `HERMES_E2E_BROWSER=1` gate that existed nowhere in
the code. That gate is now real, and the file is marked `integration`
so the default marker filter excludes it. `scripts/run_tests.sh`
forwards `HERMES_E2E_BROWSER` so the documented manual run still works.
Miscellanea
- `tests/test_audio_playback_guard.py`: regression cover for both defences,
driving the real `voice.toggle` handler rather than a stand-in.
- Move U+FFFC placeholder detection before _record_last_inbound() so the
placeholder message id is not recorded as the reaction target
- Cancel pending U+FFFC tasks in disconnect() to prevent task leaks
- Check mimeType audio/x-caf in addition to filename for CAF→VOICE promotion,
fixing unnamed attachments that default to '(unnamed)'
- Add 7 Photon adapter tests: CAF named/unnamed promotion, U+FFFC no-dispatch,
U+FFFC not recorded as last inbound, U+FFFC+attachment cancel, timeout fire,
disconnect cleanup
- Add 6 transcription tests: ffmpeg conversion, afconvert fallback, all-fail,
CAF→WAV before Groq, conversion failure error, local provider skip
_generate_xai_tts passes stream=True to requests.post since the TTS
speaker pipeline moved to the streaming core. Four fake_post mocks in
test_tts_xai_speech_tags.py still had the pre-streaming signature and
raised TypeError: unexpected keyword argument 'stream'. The other 20+
mocks in the same file already accept stream=False; this aligns the
stragglers.
Fixes the 4 failures in Python tests slice 5/8 on main.
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>
Two test classes lock in the new detection contract:
1. TestTermuxApiAppInstalledProbeLadder — drives _termux_api_app_installed
with a fake subprocess.run dispatcher and walks each rung of the
ladder:
- non-Termux env returns False (no probes run),
- pm-confirms returns True (back-compat),
- pm-clean-miss → cmd-confirms returns True,
- pm-FileNotFoundError → cmd-confirms returns True,
- pm-TimeoutExpired → cmd-confirms returns True,
- pm-nonzero-exit → cmd-confirms returns True,
- both probes inconclusive + binary on PATH returns True (the
core #31015 case),
- both probes inconclusive + no binary returns False,
- both probes clean-miss returns False (the genuine "CLI without
app" case keeps the existing warning),
- case-insensitive package match for ROMs that capitalise differently.
2. TestDetectAudioEnvironmentTermuxFallback — end-to-end through
detect_audio_environment, asserting the misleading "Termux:API
Android app is not installed" warning no longer fires when probes
are inconclusive but the binary is on PATH (the user-reported #31015
symptom), AND that the warning still fires when probes can
conclusively report 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.
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
Salvaged from PR #68063 (@malaiwah). MatrixAdapter.send_voice now transcodes
any non-Ogg audio to Ogg/Opus at the adapter boundary (best-effort — the
original file is sent unchanged when ffmpeg is unavailable), so MSC3245
voice bubbles render even when a caller hands the adapter MP3/WAV audio.
_matrix_voice_metadata_for_file probing now runs via asyncio.to_thread so
ffprobe/ffmpeg subprocess timeouts can't stall the adapter event loop.
The PR's tools/tts_tool.py want_opus hunk was dropped: main's
OPUS_VOICE_PLATFORMS set (PR #73072) already includes matrix; the Matrix
opus-routing test is kept.
Refs #14841
Consolidates all TTS text-preparation paths onto
tools/tts_text_normalize.prepare_spoken_text:
- strip_nonspoken_blocks: removes <think> reasoning blocks (#34213,
incl. unterminated streaming blocks) and the end-of-turn
file-mutation verifier footer emitted by run_agent.py (#40772).
- flatten_newlines_for_payload: collapses newlines into sentence
breaks so newline-sensitive OpenAI-compatible providers (Kokoro)
speak the whole script instead of truncating at the first newline
(#9004).
- tools/tts_tool._strip_markdown_for_tts (voice-mode streaming + web
dashboard path) now delegates to the shared cleaner, with the legacy
regex pipeline kept as a best-effort fallback.
- hermes_cli/voice.py speak_text and cli.py _voice_speak_response now
use the shared cleaner instead of their own duplicated regex
pipelines.
- gateway auto-TTS fallback also strips think blocks.
Tests: tests/tools/test_tts_prepare_spoken.py covers think blocks,
verifier footer, emoji, newline flattening, and the shared-cleaner
wiring on the tool/streaming/gateway paths. Updated the header
expectation in test_voice_cli_integration.py for the heading-fold
behavior of the shared cleaner.
Closes#34213, #9004, #40772
Address review feedback on #31693:
- Regression tests assert extra_body == {"lang_code": ...} is forwarded
when tts.openai.language is configured, and omitted when unset/empty
- Document tts.openai.language as intended for OpenAI-compatible
endpoints that support lang_code (e.g. Kokoro-FastAPI)
Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
xAI TTS supports a text_normalization boolean that normalizes
written-form text (numbers, abbreviations, symbols) into spoken-form
before synthesis. This parameter was not being sent, leaving users
with literal number/symbol pronunciation.
- Add DEFAULT_XAI_TEXT_NORMALIZATION_DEFAULT = False
- Read tts.xai.text_normalization from config (via _xai_bool_config)
- Attach text_normalization: true to the POST payload when enabled
- Omit the field entirely when unset or explicitly false (API default)
- Add 3 tests: default omission, enabled passthrough, explicit false
