- 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)
