LLM responses from browser snapshot extraction and vision analysis
could echo back secrets that appeared on screen or in page content.
Input redaction alone is insufficient — the LLM may reproduce secrets
it read from screenshots (which cannot be text-redacted).
Now redact outputs from:
- _extract_relevant_content (auxiliary LLM response)
- browser_vision (vision LLM response)
- camofox_vision (vision LLM response)
Three exfiltration vectors closed:
1. Browser URL exfil — agent could embed secrets in URL params and
navigate to attacker-controlled server. Now scans URLs for known
API key patterns before navigating (browser_navigate, web_extract).
2. Browser snapshot leak — page displaying env vars or API keys would
send secrets to auxiliary LLM via _extract_relevant_content before
run_agent.py's redaction layer sees the result. Now redacts snapshot
text before the auxiliary call.
3. Camofox annotation leak — accessibility tree text sent to vision
LLM could contain secrets visible on screen. Now redacts annotation
context before the vision call.
10 new tests covering URL blocking, snapshot redaction, and annotation
redaction for both browser and camofox backends.
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.
migrate_model_config() was writing `config["model"] = model_str` which
replaces the entire model dict (default, provider, base_url) with a
bare string. This causes 'str' object has no attribute 'get' errors
throughout Hermes when any code does model_cfg.get("default").
Now preserves the existing model dict and only updates the "default"
key, keeping provider/base_url intact.
Two bugs in the auxiliary provider auto-detection chain:
1. Expired Codex JWT blocks the auto chain: _read_codex_access_token()
returned any stored token without checking expiry, preventing fallback
to working providers. Now decodes JWT exp claim and returns None for
expired tokens.
2. Auxiliary Anthropic client missing OAuth identity transforms:
_AnthropicCompletionsAdapter always called build_anthropic_kwargs with
is_oauth=False, causing 400 errors for OAuth tokens. Now detects OAuth
tokens via _is_oauth_token() and propagates the flag through the
adapter chain.
Cherry-picked from PR #2378 by 0xbyt4. Fixed test_api_key_no_oauth_flag
to mock resolve_anthropic_token directly (env var alone was insufficient).
auth_type was "***" instead of "api_key" and api_key_env_vars was
("OPEN...",) instead of ("OPENCODE_GO_API_KEY",). This was introduced
in 35d948b6 when a secret redaction tool masked these values during
the Kilo Code provider commit. OpenCode Go provider was completely
broken as a result.
If a tool_calls list contains a None entry (from malformed API response,
compression artifact, or corrupt session replay), convert_messages_to_anthropic
crashes with AttributeError: 'NoneType' object has no attribute 'get'.
Skip None and non-dict entries in the tool_calls iteration. Found via
chaos/fuzz testing with mixed valid/invalid tool_call entries.
browser_console was registered in the tool registry but missing from
all toolset definitions (TOOLSETS, _HERMES_CORE_TOOLS, _LEGACY_TOOLSET_MAP),
so the agent could never discover or use it.
Added to all 4 locations + 4 wiring tests.
Cherry-picked from PR #1084 by @0xbyt4 (authorship preserved in tests).
Anthropic prompt caching splits input into cache_read_input_tokens,
cache_creation_input_tokens, and non-cached input_tokens. The context
counter only read input_tokens (non-cached portion), showing ~3 tokens
instead of the real ~18K total. Now includes cached portions for
Anthropic native provider only — other providers (OpenAI, OpenRouter,
Codex) already include cached tokens in their prompt_tokens field.
Before: 3/200K | 0%
After: 17.7K/200K | 9%
- 429 rate limit and 529 overloaded were incorrectly treated as
non-retryable client errors, causing immediate failure instead of
exponential backoff retry. Users hitting Anthropic rate limits got
silent failures or no response at all.
- Generic "Sorry, I encountered an unexpected error" now includes
error type, details, and status-specific hints (auth, rate limit,
overloaded).
- Failed agent with final_response=None now surfaces the actual
error message instead of returning an empty response.
Checks the full voice environment and reports what's missing:
- Python packages: discord.py, PyNaCl, davey, STT/TTS providers
- System tools: Opus codec (macOS + Linux paths), ffmpeg
- Environment: bot token, allowed users (resolved to usernames), API keys
- Configuration: STT/TTS provider, voice mode state
- Bot permissions: live Discord API check for Connect, Speak, VAD, etc.
All sensitive values are masked. Gracefully handles missing deps,
invalid tokens, API timeouts, and unreachable Discord API.
Tests cover the actual code paths changed in voice fixes:
_on_packet DAVE passthrough (8 tests):
- Known SSRC + DAVE decrypt success → buffered
- Unknown SSRC + DAVE → skip DAVE, passthrough to Opus
- DAVE "Unencrypted" error → passthrough, not dropped
- DAVE other error → packet dropped
- No DAVE session → direct decode
- Bot's own SSRC → ignored (echo prevention)
- Multiple SSRCs → separate buffers
SSRC auto-mapping (6 tests):
- Single allowed user → auto-mapped
- Multiple allowed users → no auto-map
- No allowlist → sole non-bot member inferred
- Unallowed user → rejected
- Only bot in channel → no map
- Auto-map persists across checks
Buffer lifecycle (4 tests):
- Known SSRC completed utterance
- Short buffer ignored
- Recent audio waits
- Stale unknown buffer discarded
TTS playback (10 tests):
- play_tts calls play_in_voice_channel in VC
- play_tts falls through when not in VC
- play_tts wrong channel no match
- Voice input dedup (runner skips)
- Text + voice_mode combinations
- Error/empty response skipped
- Agent TTS tool dedup
UDP keepalive (2 tests):
- Interval within bounds
- Silence frame actually sent via send_packet
When PyNaCl or davey is not installed, joining a voice channel fails
with a raw exception. Now shows a human-readable message pointing
the user to reinstall with voice support.
Closes#1336
After bot leave/rejoin, Discord doesn't resend SPEAKING events for
users already in the channel. This left SSRC unmapped and all audio
was silently dropped by DAVE decrypt.
Fixes:
- Skip DAVE for unknown SSRCs instead of dropping (passthrough)
- Handle "UnencryptedWhenPassthroughDisabled" DAVE errors gracefully
- Auto-infer user_id from sole allowed member in voice channel
- Pass allowed_user_ids to VoiceReceiver for secure inference
Discord drops the UDP voice route after ~60s of silence - no packets
arrive even when users start speaking again. Send an Opus silence
frame every 15s to keep the UDP session alive.
play_tts was returning success without playing anything when bot was
in a voice channel. Now it calls play_in_voice_channel directly.
Simplified skip_double dedup: base adapter handles voice input TTS
via play_tts (which now works for VC), runner skips to avoid double.
pattern_key was derived by splitting the regex on \b and taking [1],
so patterns starting with the same word (e.g. find -exec rm and
find -delete) produced the same key "find". Approving one silently
approved the other. Using the unique description string as the key
eliminates all collisions.
The fork bomb regex used `()` (empty capture group) and unescaped `{}`
instead of literal `\(\)` and `\{\}`. This meant the classic fork bomb
`:(){ :|:& };:` was never detected. Also added `\s*` between `:` and
`&` and between `;` and trailing `:` to catch whitespace variants.
When the Responses API returns tool call arguments as a dict,
str(dict) produces Python repr with single quotes (e.g. {'key': 'val'})
which is invalid JSON. Downstream json.loads() fails silently and the
tool gets called with empty arguments, losing all parameters.
Affects both function_call and custom_tool_call item types in
_normalize_codex_response().
Moonshot (legacy key) users were shown kimi-for-coding and
kimi-k2-thinking-turbo which only work on the Coding Plan endpoint
(api.kimi.com/coding/v1). Add a separate "moonshot" model list that
excludes plan-specific models.
The rebase added voice prompt checks to _get_tui_prompt_fragments but
the test stub was missing _voice_recording, _voice_processing and
_voice_mode attributes, causing AttributeError.
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.
The mock's app_commands SimpleNamespace lacked choices and Choice attrs,
causing xdist test ordering failures when this mock loaded before
test_discord_slash_commands.
1. Gate _streaming_api_call to chat_completions mode only — Anthropic and
Codex fall back to _interruptible_api_call. Preserve Anthropic base_url
across all client rebuild paths (interrupt, fallback, 401 refresh).
2. Discord VC synthetic events now use chat_type="channel" instead of
defaulting to "dm" — prevents session bleed into DM context.
Authorization runs before echoing transcript. Sanitize @everyone/@here
in voice transcripts.
3. CLI voice prefix ("[Voice input...]") is now API-call-local only —
stripped from returned history so it never persists to session DB or
resumed sessions.
4. /voice off now disables base adapter auto-TTS via _auto_tts_disabled_chats
set — voice input no longer triggers TTS when voice mode is off.
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.
- web.py: pass stt_model from config like discord.py and run.py do
- run.py: match new error messages (No STT provider / not set)
- _transcribe_local: add missing "provider": "local" to return dict
Move stream close outside the lock in shutdown() to prevent deadlock
when audio callback tries to acquire the same lock. Replace single
t.join(timeout) with a polling loop (0.1s intervals) so KeyboardInterrupt
is not blocked during stream cleanup.
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".
Merge main's faster-whisper (local, free) with our Groq support into a
unified three-provider STT pipeline: local > groq > openai.
Provider priority ensures free options are tried first. Each provider
has its own transcriber function with model auto-correction, env-
overridable endpoints, and proper error handling.
74 tests cover the full provider matrix, fallback chains, model
correction, config loading, validation edge cases, and dispatch.
- Change RTP packet logging from INFO to DEBUG level to reduce noise
(SPEAKING events remain at INFO as they are important lifecycle events)
- Use per-session chat_id (web_{session_id}) instead of shared "web"
to isolate conversation context between simultaneous web users
When bound to 127.0.0.1, only show localhost URL instead of listing
unreachable network interfaces. Add hint about WEB_UI_HOST=0.0.0.0
for phone/tablet access. Add VPN/multi-interface and token exposure
tests (11 new tests).
Only print the access token when auto-generated (user needs it to
log in). When set via WEB_UI_TOKEN env var, just confirm it is set
without exposing the value in console output.
- Path traversal sanitization (Path.name strips ../)
- Media endpoint authentication (401 without token, 404 on traversal)
- hmac.compare_digest usage verification (no == for tokens)
- DOMPurify XSS prevention in HTML template
- Default bind 127.0.0.1 (adapter and config)
- /remote-control token hiding in group chats
- Opus find_library instead of hardcoded paths
- Opus decode error logging (no silent swallow)
- Interrupt _vprint force=True on all 6 calls
- Anthropic interrupt handler in both API call paths
- Update test_web_defaults for new 127.0.0.1 default
- Use hmac.compare_digest for timing-safe token comparison (3 endpoints)
- Default bind to 127.0.0.1 instead of 0.0.0.0
- Sanitize upload filenames with Path.name to prevent path traversal
- Add DOMPurify to sanitize marked.parse() output against XSS
- Replace add_static with authenticated media handler
- Hide token in group chats for /remote-control command
- Use ctypes.util.find_library for Opus instead of hardcoded paths
- Add force=True to 5 interrupt _vprint calls for visibility
- Log Opus decode errors and voice restart failures instead of swallowing
Rebase auto-merge silently overwrote main's Anthropic-aware interrupt
handler with the older OpenAI-only version. Without this fix, interrupting
an Anthropic API call closes the wrong client and leaves token generation
running on the Anthropic side.
Duplicated YAML config parsing for stt.model existed in gateway/run.py
and gateway/platforms/discord.py. Moved to a single helper in
transcription_tools.py and added 5 tests covering all edge cases.
Code fixes:
- STT model, Groq base URL, and OpenAI STT base URL are now
configurable via env vars (STT_GROQ_MODEL, STT_OPENAI_MODEL,
GROQ_BASE_URL, STT_OPENAI_BASE_URL) instead of hardcoded
- Gateway and Discord VC now read stt.model from config.yaml
(previously only CLI did this — gateway always used defaults)
Doc fixes:
- voice-mode.md: move Web UI troubleshooting to web.md (was duplicated)
- voice-mode.md: simplify "How It Works" for end users (remove NaCl,
DAVE, RTP internals)
- voice-mode.md: clarify STT priority (OpenAI used first if both keys
set, Groq recommended for free tier)
- voice-mode.md: document new STT env overrides in config reference
- web.md: remove duplicate Quick Start / Step 1-3 sections
- web.md: add mobile HTTPS mic workarounds (moved from voice-mode.md)
- web.md: clarify STT fallback order
- voice-mode.md: Discord sends native voice bubbles (OGG/Opus flags=8192),
not MP3 file attachments. Falls back to file only if voice API fails.
- discord.md: Bot requires @mention by default in server channels
(DISCORD_REQUIRE_MENTION=true). Previous text incorrectly said no
mention needed.
- index.md: Fix broken ASCII architecture diagram alignment after
adding Web adapter box.
1. VoiceReceiver.stop() now acquires _lock before clearing shared state
to prevent race with _on_packet on the socket reader thread
2. _packet_debug_count moved from class-level to instance-level to avoid
cross-instance race condition in multi-guild setups
3. play_in_voice_channel uses asyncio.get_running_loop() instead of
deprecated asyncio.get_event_loop()
4. _send_voice_reply uses uuid for filenames instead of time-based names
that can collide when two replies happen in the same second
5. Voice timeout now notifies runner via _on_voice_disconnect callback
so runner cleans up _voice_mode state (prevents orphaned TTS replies)
6. play_in_voice_channel adds PLAYBACK_TIMEOUT (120s) to prevent
infinite blocking when FFmpeg callback is never called
7. _send_voice_reply moves temp file cleanup to finally block so files
are always cleaned up even when send_voice/play raises
8. Base adapter auto-TTS wraps play_tts in try/finally with os.remove
to clean up generated audio files after playback
18 new tests (120 total voice tests)
- Add lock protection around VoiceReceiver buffer writes in _on_packet
to prevent race condition with check_silence on different threads
- Wire _voice_input_callback BEFORE join_voice_channel to avoid
losing voice input during the join window
- Add try/except around leave_voice_channel to ensure state cleanup
(voice_mode, callback) even if leave raises an exception
- Guard against empty text after markdown stripping in base.py auto-TTS
- Add 11 tests proving each bug and verifying the fix
- Import from tools.tts_tool instead of reimplementing the logic
- Fix test_truncates_long_text: truncation is the caller's job, not the function's
- Remove unused re import
- macOS firewall may block LAN access to Web UI
- Mobile browsers require HTTPS for microphone API
- Document workarounds: Android Chrome flag, mkcert self-signed cert,
Caddy reverse proxy, SSH tunnel for iOS
Mobile browsers require HTTPS for navigator.mediaDevices API.
Instead of hiding the mic button (confusing UX), show it as dimmed
and display an informative message when tapped explaining the HTTPS
requirement.
When bot is in a Discord voice channel, both base auto-TTS and Discord
play_tts override skip audio. The skip_double guard was also blocking
the runner's _send_voice_reply, resulting in zero audio output in VC.
Now skip_double is overridden when the bot is actively connected to a
voice channel, allowing play_in_voice_channel to handle TTS.
Add comprehensive test matrix covering all platform x input x mode
combinations with full decision table documentation.
- Update TestAutoVoiceReply to include skip_double logic: voice input
is handled by base adapter auto-TTS, gateway runner skips to prevent
duplicate audio
- Add TestDiscordPlayTtsSkip: verifies Discord adapter skips play_tts
when bot is in a voice channel (VC playback handled by runner)
- Add TestWebPlayTts: verifies Web adapter sends invisible play_audio
instead of voice bubble
Base adapter auto-TTS already generates and sends audio for voice
messages in _process_message_background. The gateway runner's
_send_voice_reply was causing double audio on all platforms (not
just Web). Now skip_double applies to any voice input regardless
of platform.
Override play_tts in DiscordAdapter to no-op when connected to a voice
channel for the same guild. The gateway runner already plays TTS audio
in the VC via play_in_voice_channel, so the base adapter's fallback
to send_voice (file attachment) was causing double audio output.
When voice mode is enabled and user sends a voice message on Web UI,
both the base adapter auto-TTS (play_audio) and the gateway voice reply
(send_voice) would fire, causing duplicate audio playback. Skip the
gateway voice reply for Web platform voice input since base adapter
already handles it.
- Document DM vs server channel interaction modes
- Explain @mention requirement and how to select bot user vs role
- Add DISCORD_REQUIRE_MENTION and DISCORD_FREE_RESPONSE_CHANNELS config
- Add troubleshooting entry for bot not responding in server channels
play_tts base class forwards metadata via **kwargs to send_voice,
but Discord and Slack adapters did not accept extra keyword arguments,
causing TypeError and silent message handling failure.
Also fix test_web_defaults to patch correct env var (WEB_UI_TOKEN).
- Voice mode: press mic once to enter, press again to exit
- VAD (Voice Activity Detection) auto-stops recording after 1.5s silence
- Continuous loop: speak → transcribe → agent responds → TTS plays → auto-listen
- Voice mode UI: input bar hides, large mic button centered
- Auto-restart listening when TTS playback finishes
- Fallback: restart listening on text response if no TTS arrives
- Auto-TTS: voice messages get spoken response (audio first, then text)
- STT: Groq Whisper fallback when VOICE_TOOLS_OPENAI_KEY not set
- Futuristic UI: glassmorphism, centered container, purple theme, glow effects
- Voice bubble: custom waveform player with seek and progress
- Invisible TTS playback via play_tts() method (no audio file in chat)
- Add hermes-web toolset with full tool access
- Register Platform.WEB in toolset/config maps
- Update docs for voice conversation feature
Detect all network interfaces instead of relying on UDP trick which
returns VPN IP. Prefers 192.168.x.x/10.x.x.x over VPN ranges.
Shows all available IPs in console output.
Type /remote-control from any platform (Telegram, Discord, etc.) to
instantly start the web UI without restarting the gateway.
- Auto-generates access token if not provided
- Shows URL + token in response
- Optional: /remote-control [port] [token]
- Reports status if already running
- Added to /help command list
New platform adapter that serves a full-featured chat interface via HTTP.
Enables access from any device on the network (phone, tablet, desktop).
Features:
- aiohttp server with WebSocket real-time messaging
- Token-based authentication
- Markdown rendering (marked.js) + code highlighting (highlight.js)
- Voice recording via MediaRecorder API + STT transcription
- Image, voice, and document display
- Typing indicator + message editing (streaming support)
- Mobile responsive dark theme
- Auto-reconnect on disconnect
- Media file cleanup (24h TTL)
Config: WEB_UI_ENABLED=true, WEB_UI_PORT=8765, WEB_UI_TOKEN=<token>
No new dependencies — uses aiohttp already in [messaging] extra.
Cover CLI voice mode, Telegram/Discord auto voice reply, and Discord
voice channel support. Include setup guide with bot permissions, OAuth2
invite URL, privileged intents, system dependencies, and Python packages.
Update discord.md voice messages section with correct STT key reference.
Phase 2 of voice channel support: bot listens to users speaking in VC,
transcribes speech via Groq Whisper, and processes through the agent pipeline.
- Add VoiceReceiver class for RTP packet capture, NaCl/DAVE decryption, Opus decode
- Add silence detection and per-user PCM buffering
- Wire voice input callback from adapter to GatewayRunner
- Fix adapter dict key: use Platform.DISCORD enum instead of string
- Fix guild_id extraction for synthetic voice events via SimpleNamespace raw_message
- Pause/resume receiver during TTS playback to prevent echo
- Send Discord voice messages with flags=8192 and waveform metadata
so they render as native voice bubbles instead of file attachments
- Use .mp3 output path for TTS so edge-tts opus conversion works
correctly (edge always outputs mp3, convert was skipped for .ogg)
- Use actual file_path from TTS result after potential opus conversion
- Register /voice as Discord slash command with mode choices
- Fix _send_voice_reply to handle adapters that don't accept metadata
parameter (Discord) by inspecting the method signature at runtime
- /voice on: reply with voice when user sends voice messages
- /voice tts: reply with voice to all messages
- /voice off: disable, text-only replies
- /voice status: show current mode
- Per-chat state persisted to gateway_voice_mode.json
- Dedup: skips auto-reply if agent already called text_to_speech tool
- drop_pending_updates=True to ignore stale Telegram messages on restart
- 25 tests covering command handler, reply logic, and edge cases
The counter was incremented in start/stop/cancel but never read
anywhere in the codebase. The race condition it was meant to guard
against is practically impossible with the persistent stream design.
- 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.
AudioRecorder now auto-stops after 15 seconds if no speech is detected
(_has_spoken remains False). In quiet environments where ambient RMS
never exceeds the silence threshold (200), the recording would wait
indefinitely. The new _max_wait parameter fires the silence callback
after the timeout, triggering the normal "No speech detected" flow.
- Set max_retries=0 on the STT OpenAI client. The SDK default (2) honors
Groq's retry-after header (often 53s), blocking the thread for up to
~106s on rate limits. Voice STT should fail fast, not retry silently.
- Stop continuous recording mode after 3 consecutive no-speech cycles to
prevent infinite restart loops when nobody is talking.
- Set OpenAI client timeout=30s in transcribe_audio() — default 600s
blocks _voice_processing for 10 min if Groq/OpenAI stalls
- Move _voice_start_recording in _voice_stop_and_transcribe finally
block to a daemon thread (same pattern as Ctrl+B handler and
process_loop)
- Add _should_exit guard at top of _voice_start_recording so all 4
call sites respect shutdown without individual checks
- Replace sd.wait() with a poll loop + sd.stop() in play_beep().
sd.wait() calls Event.wait() without timeout — hangs forever if the
audio device stalls. Poll with a 2s ceiling and force-stop instead.
- Wrap _on_silence callback in try-except so exceptions are logged
instead of silently lost in the daemon thread. Prevents recording
state from becoming inconsistent on unexpected errors.
- process_loop's continuous mode restart called _voice_start_recording()
directly, blocking the loop if play_beep/sd.wait hangs — queued user
input would stall silently. Dispatch to daemon thread like Ctrl+B handler.
- Replace print() with _cprint() in _handle_voice_command for consistency
with the rest of the voice mode code.
The handle_voice_record key binding runs in prompt_toolkit's event-loop
thread. When silence auto-stopped recording, _voice_recording was False
but recorder.stop() still held AudioRecorder._lock. A concurrent Ctrl+B
press entered the START path and blocked on that lock, freezing all
keyboard input.
Three changes:
- Set _voice_processing atomically with _voice_recording=False in
_voice_stop_and_transcribe to close the race window
- Add _voice_processing guard in the START path to prevent starting
while stop/transcribe is still running
- Dispatch _voice_start_recording to a daemon thread so play_beep
(sd.wait) and AudioRecorder.start (lock acquire) never block the
event loop
browser_tool.py registered SIGINT/SIGTERM handlers that called sys.exit()
at module import time. When a signal arrived during a lock acquisition
(e.g. AudioRecorder._lock in voice mode), SystemExit was raised inside
prompt_toolkit's async event loop, corrupting coroutine state and making
the process unkillable (required SIGKILL).
atexit handler already ensures browser sessions are cleaned up on any
normal exit path, so the signal handlers were redundant and harmful.
- edge_tts NameError: _generate_edge_tts now calls _import_edge_tts()
instead of referencing bare module name (tts_tool.py)
- TTS thread leak: chat() finally block sends sentinel to text_queue,
sets stop_event, and joins tts_thread on exception paths (cli.py)
- output_stream leak: moved close() into finally block so audio device
is released even on exception (tts_tool.py)
- Ctrl+C continuous mode: cancel handler now resets _voice_continuous
to prevent auto-restart after user cancels recording (cli.py)
- _disable_voice_mode: now calls stop_playback() and sets
_voice_tts_done so TTS stops when voice mode is turned off (cli.py)
- _show_voice_status: reads record key from config instead of
hardcoding Ctrl+B (cli.py)
Bug A: Replace stale _HAS_ELEVENLABS/_HAS_AUDIO boolean imports with
lazy import function calls (_import_elevenlabs, _import_sounddevice).
The old constants no longer exist in tts_tool -- the try/except
silently swallowed the ImportError, leaving streaming TTS dead.
Bug B: Use user message prefix instead of modifying system prompt for
voice mode instruction. Changing ephemeral_system_prompt mid-session
invalidates the prompt cache. Now the concise-response hint is
prepended to the user_message passed to run_conversation while
conversation_history keeps the original text.
Minor: Add force parameter to _vprint so critical error messages
(max retries, non-retryable errors, API failures) are always shown
even during streaming TTS playback.
Tests: 15 new tests in test_voice_cli_integration.py covering all
three fixes -- lazy import activation, message prefix behavior,
history cleanliness, system prompt stability, and AST verification
that all critical _vprint calls use force=True.
- AudioRecorder.start() now catches InputStream errors gracefully
with a clear error message about microphone availability
- Fix config key mismatch: cli.py was reading "push_to_talk_key"
but config.py defines "record_key" -- now consistent
- Add format conversion from config format ("ctrl+b") to
prompt_toolkit format ("c-b")
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.
- Fix Gemini streaming tool call merge bug: multiple tool calls with same
index but different IDs are now parsed as separate calls instead of
concatenating names (e.g. ha_call_serviceha_call_service)
- Handle partial results in voice mode: show error and stop continuous
mode when agent returns partial/failed results with empty response
- Fix error display during streaming TTS: error messages are shown in
full response box even when streaming box was already opened
- Add duplicate sentence filter in TTS: skip near-duplicate sentences
from LLM repetition
- Fix fake HA server state mutation: turn_on/turn_off/set_temperature
correctly update entity states; temperature sensor simulates change
when thermostat is adjusted
- Add _vprint() helper to suppress log output when stream_callback is active
- Expand Whisper hallucination filter with multi-language phrases and regex pattern for repetitive text
- Stop continuous voice mode when agent returns a failed result (e.g. 429 rate limit)
- Atomic check-and-set for _voice_recording flag with _voice_lock
- Guard _voice_stop_and_transcribe against concurrent invocation
- Remove premature flag clearing from Ctrl+R handler
- Clean up temp WAV files in finally block (_play_via_tempfile)
- Use buffer-level regex for <think> block filtering (handles chunked tags)
- Prevent /voice on prompt accumulation on repeated calls
- Include Groq in STT key error message
0xbyt4