ThinkingConfig was defined as an inner class on the service but referenced in the Settings dataclass declared before the service class, causing a crash at import time. Move ThinkingConfig to a standalone class defined before Settings, and keep a class attribute alias for backward compatibility.
Eliminate custom _emit_stt_ttfb_metric and manual timestamp tracking in
STTService by reusing FrameProcessor's start_ttfb_metrics/stop_ttfb_metrics
with new start_time/end_time parameters. This keeps the chronological
start→stop ordering and removes _speech_end_time and _last_transcription_time
state from STTService.
Remove the deprecation warning and __post_init__ override. Also fix the
default value for remote_participants to use field(default_factory=dict)
instead of None.
Add write_transport_frame() hook to BaseOutputTransport so subclasses
can handle custom frame types that flow through the audio queue. Add
DailySIPTransferFrame and DailySIPReferFrame as DataFrame subclasses
that queue with audio, ensuring SIP operations execute only after the
bot finishes its current utterance. Override write_transport_frame in
DailyOutputTransport to dispatch these frames to the existing
sip_call_transfer() and sip_refer() client methods.
Also switch DailyOutputTransport.send_message error handling from
logger.error to push_error for consistency.
Every `*Settings` dataclass field whose default is `NOT_GIVEN` now carries `_NotGiven` in its type union so the type system accurately reflects the three-state semantics (real value, `None` where applicable, or not-yet-specified). Fields previously typed as bare `Any`, `str`, `float`, `bool`, `list`, `dict`, or `Optional[X]` are now narrowed to the specific type from the corresponding `InputParams` Pydantic model.
RTVIObserver previously filtered out all upstream frames to avoid
duplicate messages from broadcasted frames. This caused upstream-only
frames to be silently ignored. Instead, add a `broadcasted` field to
the Frame base class that is set by broadcast_frame() and
broadcast_frame_instance(), and only skip upstream copies of
broadcasted frames.
- Move `CommitStrategy` up in the file so it could be used by `ElevenLabsRealtimeSTTSettings`
- Fix a bug where `run_tts` would erroneously try to reconnect if a reconnection was already in flight (like a reconnection triggered by `_update_settings`)
HumeTTSService now stores its params (description, speed, trailing_silence) in a proper `HumeTTSSettings` dataclass instead of a separate `_params` Pydantic model, making it work with `TTSUpdateSettingsFrame(update=...)`. The old `update_setting(key, value)` method is kept but deprecated.
Also removes the unused no-op `TTSService.update_setting` base method, which was never called by the `TTSUpdateSettingsFrame` pipeline.
The dataclass-based API (`*UpdateSettingsFrame(update=*Settings(...))`) is the preferred path since 0.0.103. The dict path still works but now emits a `DeprecationWarning`.
Change `TTSSettings.language` and `STTSettings.language` from `Any` to `Language | str | _NotGiven`. Add `language_to_service_language` base method and centralized `isinstance`-guarded conversion in `STTService._update_settings` (mirroring TTS). Update the TTS guard from `is not None` to `isinstance(…, Language)` so raw strings pass through unchanged.
Remove now-redundant per-service language conversion from `_update_settings` overrides (ElevenLabs, Azure, Fal, Whisper). Add `language_to_service_language` to Azure STT so the centralized conversion picks it up. Fix AWS and NVIDIA STT `__init__` to convert language at construction time, then simplify their runtime accessors to read `_settings.language` directly.
Note that for services that previously handled applying updates (through methods like `set_model` and `set_language`), we're keeping the update-applying logic (some or most of which is already well-tested) and expanding it to cover all relevant settings fields. Services under this bucket are:
- Deepgram STT
- Deepgram Sagemaker STT
- Elevenlabs STT
- Google STT
- Gradium STT
- OpenAI STT
- Speechmatics STT
Use client_req_id-based multiplexing instead of disconnecting and
reconnecting the websocket on every interruption. This follows the
same pattern used by Cartesia, ElevenLabs, and other services via
AudioContextWordTTSService.
Key changes:
- Base class: InterruptibleWordTTSService -> AudioContextWordTTSService
- Add close_ws_on_eos: False to setup message to keep connection alive
- Add client_req_id to text, end_of_stream messages for demultiplexing
- Route audio via append_to_audio_context() instead of push_frame()
- Silently drop messages for cancelled/unknown contexts on interruption
- Add _handle_interruption() that resets context without reconnecting
- Remove no-op push_frame() override
