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".
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.
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).
- 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
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.
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
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
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).
- 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
- 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.
- 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.
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
- 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.
Salvaged from PR #932 by Wayne onto current main.
Apply skin-aware prompt symbols and live prompt_toolkit color refresh,
replace lingering hardcoded accent output with active-skin colors, keep
ANSI-safe response rendering, preserve secret-capture and approval-prompt
state handling, and add integration coverage for prompt state and style
refresh behavior.
* feat: improve context compaction handoff summaries
Adapt PR #916 onto current main by replacing the old context summary marker
with a clearer handoff wrapper, updating the summarization prompt for
resume-oriented summaries, and preserving the current call_llm-based
compression path.
* fix: clearer error when docker backend is unavailable
* fix: preserve docker discovery in backend preflight
Follow up on salvaged PR #940 by reusing find_docker() during the new
availability check so non-PATH Docker Desktop installs still work. Add
a regression test covering the resolved executable path.
* test: make gateway async tests xdist-safe
Replace sync test usage of asyncio.get_event_loop().run_until_complete()
with asyncio.run() so tests do not depend on an ambient current event loop.
Also create the email disconnect poll task inside a running loop. This fixes
xdist/CI failures where workers have no current loop in MainThread.
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Co-authored-by: aydnOktay <xaydinoktay@gmail.com>
