Salvage of PR #52188. The original PR raised RuntimeError when LM Studio
load was rejected or unverifiable, which would abort agent startup on
transient network failures. Replace with logger.warning + fallback to
configured context length, preserving the old graceful-degradation behavior.
Bare-login noreply emails (no NNN+ numeric prefix) do not match the
check-attribution auto-resolve rule, so they need an explicit
contributors/emails/ file. Unblocks PR #73592.
* fix(desktop): allow camera capture through the permission handlers
The session permission hooks were written for the voice composer and denied
video outright, so any renderer getUserMedia({video}) failed with
NotAllowedError before the OS was ever consulted.
Rename isAudioCapturePermission to isMediaCapturePermission and accept video
alongside audio in both the request and check handlers. The OS capture
permission still applies, so the user keeps a real allow/deny.
* build(desktop): declare camera usage for signed macOS builds
A hardened-runtime build needs the camera entitlement and an
NSCameraUsageDescription string, or the packaged app is killed on first
camera access instead of prompting.
_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.
The codex app-server namespaces MCP tool call ids as
codex_mcp__<server>__<tool>_<codex_call_id>. With an exec-<uuid> component the
built-in hermes-tools server alone overflows the Responses API's 64-char
call_id limit, so the request 400s with a non-retryable "string too long".
The offending item sits near the head of the transcript and replays every
turn, permanently bricking the session — the only recovery is /reset.
Sibling defect to #10788, which clamped input[*].id via
_MAX_RESPONSES_ITEM_ID_LENGTH. Apply the same treatment to call_id at both
Responses emit sites in _chat_messages_to_responses_input: a deterministic
surrogate (call_ + sha256[:32]) for ids over the limit, short ids unchanged.
Because the surrogate is a pure function of the original id, a function_call
and its matching function_call_output — which carry the same original id — map
to the same surrogate and stay paired without correlating the two items.
Two log-spam bugs found in live gateway logs:
1. gateway_routing UNIQUE-constraint spam (261 warnings in one errors.log):
early builds of the #59203 routing-index migration created
gateway_routing with 'session_key TEXT PRIMARY KEY' and no scope
column. _reconcile_columns() ADDs the missing scope column but SQLite
cannot ALTER a primary key, so the shipped composite
PRIMARY KEY (scope, session_key) never lands on those databases. Both
write paths then fail on every save:
- save_gateway_routing_entry: 'ON CONFLICT clause does not match any
PRIMARY KEY or UNIQUE constraint'
- replace_gateway_routing_entries: 'UNIQUE constraint failed:
gateway_routing.session_key' whenever the same session_key exists
under another scope (e.g. test-suite scopes leaked into a live DB).
New _heal_gateway_routing_pk() rebuilds the table once with the
composite key, preserving rows (newest wins on collisions, NULL scope
coalesced to ''). Same one-time-heal pattern as the #51646 active-
column repair. Verified E2E against a copy of a real affected state.db.
2. pre_gateway_dispatch warned ''GatewayRunner' object has no attribute
'session_store'' and silently dropped the hook for every message on
partially-initialized runners (bare object.__new__ runners in tests,
and any future init-order change). Pass
getattr(self, 'session_store', None) so the hook always fires
(pitfall #17 pattern).
Both regression tests fail without their fixes (sabotage-verified).
The /api/audio/speak-stream WebSocket URL was always minted for the primary
backend with no profile param, so streaming read-aloud used the default
profile's TTS provider even when another profile was active. Mint the
ticket for the active profile's connection and carry ?profile= (same seam
as /api/pty), via a read-only getApiRequestProfile() accessor.
/api/audio/transcribe, /api/audio/speak, /api/audio/elevenlabs/voices, and
the /api/audio/speak-stream WebSocket resolved TTS/STT config from the
dashboard's own HERMES_HOME regardless of the active profile, so a
non-default profile's voice settings were silently ignored. Give all four
the same optional profile param as the rest of the dashboard surface,
entering _config_profile_scope (await-safe, config-only — the audio paths
touch no skills globals) inside their worker threads.
Backend half of the desktop fix; completes the renderer-side profileScoped()
threading. Fixes#53441#45506#66012#64057.
Generated media often arrives as image markdown (). The chat
markdown renderer maps the img element to MarkdownImage, which renders a raw
<img>; for a video/audio source the browser cannot paint it and shows a
broken-image icon even though the file is valid and plays fine. Images worked
only because <img src=...png> is valid markup.
Route video/audio sources in MarkdownImage to the existing MediaAttachment,
which already picks the correct <video>/<audio> element (streaming protocol +
open-externally fallback) by media kind. The intended MEDIA:-tag -> #media:
link path is unaffected; this only fixes the bare image-markdown case.
Since #57944 the image path is built on hooks (async src resolution, failure
and loading states), so the routing check cannot be a plain early return
inside it -- it would have to sit after every hook call and would still fire
an image resolve for media never rendered as an image. MarkdownImage is
therefore a thin hookless router in front of MarkdownImageContent, which is
the previous body unchanged.
The regression tests join the existing markdown-text.media.test.tsx added by
#57944 rather than replacing it.
Fixes#40896
Co-Authored-By: Claude Opus 5 (1M context) <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_018vU45SEvcxLkVeyyEE9gJ2
transcribeAudio/speakText/getElevenLabsVoices called the backend without
profileScoped(), so for a non-default profile they hit the default backend
config instead of the active profile - playback used the wrong TTS/voice
even though settings were saved to the active profile config.
Fixes#53441
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
Typed messages sent while voice mode was active were also getting the
'[Voice input — respond concisely...]' API-local prefix, because the gate
checked only self._voice_mode. Route STT transcripts through a
_VoiceInputMessage sentinel in _pending_input (both the PTT/continuous
transcription path and the barge-in utterance path), unwrap it in
process_loop, and thread voice_input= through chat() so the prefix applies
only to genuinely voice-transcribed messages.
Re-cut of PR #65961 (@webtecnica) — the original diff had the sentinel
class embedded inside __init__'s docstring. Credit also to the earliest
route-by-origin attempt in PR #11744 (@KeroZelvin).
Fixes#65827Closes#65961Closes#11744
The quickstart voice-mode section used `uv pip install -e ".[voice]"`
which fails on a fresh curl-installed setup because no virtual environment
is active. Use `--python ./venv/bin/python` to target the installer-created
venv explicitly, matching the curl installer layout.
Fixes#44364
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>
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
\`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.
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
When voice continuous mode detects 3 consecutive no-speech cycles it sets
_voice_continuous = False and previously did an immediate
eturn. The
restart guard if self._voice_continuous and not submitted and not
self._voice_recording came *after* the early return, so the thread could
never restart. However a timing window existed: if _voice_start_recording was
already queued from a prior iteration, the
eturn bypassed the guard while
the thread was already in flight.
Replace the bare
eturn with a stop_continuous_restart boolean evaluated
in the same if that guards _restart_recording. This ensures both branches
read a consistent no-restart signal and no recording thread is spawned after
the user's session has been intentionally halted.
In continuous voice mode, pressing the record hotkey (Ctrl+B) was a
silent no-op while the agent was running or voice was being transcribed.
The hotkey only cleared _voice_continuous when _voice_recording was True,
leaving the user trapped in an auto-restart loop that only /voice off
could break.
Fix: when the agent is running or transcribing, still allow the hotkey
to clear _voice_continuous so the loop stops after the current turn.
Follow-up for salvaged #65023/#53020: _prepare_busy_steer_text now calls
_transcribe_and_echo_pending_voice (the same helper the interrupt monitor
and pending-drain paths use) instead of a private transcription+echo copy,
so out-of-band voice pays one STT call per platform message and the echo
respects the count-based ledger from #67281. can_steer now accepts events
whose attachments are all STT-eligible voice media, completing the steer
half of #58780. Adds extract_media gating tests for #44826 and the
contributor mapping for chefboyrdave21.
_invalidate_pending_stt_cache() clears the gateway-side transcription
cache when merge_pending_message_event() folds a follow-up message into a
still-pending event, so the next transcription picks up the merged text
and attachments. It also cleared _gateway_pending_stt_echo_sent, but that
flag is not derived state — it records that the transcript was already
delivered to the user.
Dropping it makes the re-run transcription echo the earlier notes a second
time. Both merge branches are affected, including the text-only follow-up
case where no new audio arrived at all: there the cache is invalidated,
the same voice note is transcribed again (a second paid STT call) and the
same line is echoed again.
Sequence:
1. voice note arrives, interrupt monitor transcribes it and echoes
'🎙️ "hello"'
2. user sends a follow-up while the turn is still pending, so it merges
3. drain path re-transcribes and echoes '🎙️ "hello"' a second time
Keep the ledger out of the invalidation set and track it as a count of
already-echoed transcripts instead of a single boolean. A count is what
the merge case actually needs: re-running transcription over the extended
media list returns the earlier transcripts as a prefix of the new one, so
echoing only the unsent tail suppresses the repeat while still surfacing a
newly merged voice note. A count rather than a set of seen values, so two
separate notes that transcribe identically stay two distinct deliveries —
covered by test_pending_stt_merge_echoes_two_identical_transcripts.
The guard stays within the 12-line window that
test_all_gateway_transcript_echo_sends_are_gated enforces over run.py.
