* Remove duplicate reconnection logic from Gradium STT
The _receive_messages method had its own while-True reconnect loop,
duplicating the reconnection handling already provided by
WebsocketService._receive_task_handler (exponential backoff, max
retries, error reporting). Flatten to just the inner message loop
and let the base class handle reconnection.
* Align Gradium STT VAD handling with base class patterns
Replace the process_frame override with a _handle_vad_user_stopped_speaking
override, which is the proper hook provided by STTService. Move
start_processing_metrics() into run_stt (matching Gladia's pattern).
Remove unused FrameDirection and VADUserStartedSpeakingFrame imports.
* Add transcript aggregation delay after flushed to capture trailing tokens
Gradium flushed response can arrive before all text tokens have been
delivered. Instead of finalizing immediately on flushed, start a short
timer (100ms) that allows trailing tokens to accumulate before pushing
the final TranscriptionFrame.
* Add changelog for PR #4066
* Change default encoding to pcm_16000
* Decouple encoding from sample_rate in Gradium STT
The encoding parameter now takes just the base type (pcm, wav, opus)
and the sample rate is derived from the pipeline audio_in_sample_rate,
assembled dynamically via input_format_from_encoding(). This fixes the
mismatch where SAMPLE_RATE=24000 was passed to the base class while
encoding defaulted to pcm_16000.
List-valued settings like keyterm, keywords, search, redact, and replace
were being converted to strings before being passed to the SDK connect()
method. The SDK expects lists so its encode_query can produce repeated
query params (keyterm=a&keyterm=b).
Expose enable_dialout as a configure() parameter (default False) so
dial-out examples can opt in without needing to build DailyRoomProperties
manually.
Narrow misleading Optional type hints on parameters that never accept
None, extract the duplicated token_exp_duration * 60 * 60 calculation,
remove unnecessary forward-reference quotes on DailyMeetingTokenProperties,
and clarify why enable_dialout is explicitly set to False.
Refactor language_to_soniox_language to use resolve_language + LANGUAGE_MAP
pattern consistent with other services. Fix resolve_language fallback to use
str(language) instead of language.value so plain strings don't crash.
The Inworld WS TTS plugin previously relied on the base TTS service's 3-second AUDIO_CONTEXT_TIMEOUT to detect when audio was done, then sent close_context in on_audio_context_completed. This added unnecessary latency before TTSStoppedFrame was emitted.
The original implementation likely borrowed this idea from the 11labs' impelementation. But it's likely better to mirror the Cartesia plugin pattern where on_audio_context_completed is a no-op because the server signals completion directly.
Now close_context is sent in on_turn_context_completed (right after flush_context), so the server responds with contextClosed immediately after the last audio byte. The existing receive handler already calls remove_audio_context on contextClosed, which exits the audio context handler cleanly.
The default model for OpenAILLMService and AzureLLMService was still set
to gpt-4o. Restored it to gpt-4.1. Also, removed hardcoded gpt-4o/gpt-4o-mini
model references from examples so they pick up the new default.
* Add ServiceSwitcherStrategyFailover for automatic error-based service switching
Introduce a strategy hierarchy: ServiceSwitcherStrategy (base) →
ServiceSwitcherStrategyManual (handles ManuallySwitchServiceFrame) →
ServiceSwitcherStrategyFailover (adds error-based failover). ServiceSwitcher
now defaults to ServiceSwitcherStrategyManual with strategy_type optional.
Non-fatal ErrorFrames are forwarded to the strategy via handle_error().
* Move metadata request into _set_active_if_available
Requesting metadata is part of making a service active, so it belongs
alongside setting _active_service and firing on_service_switched. This
removes the duplicate queue_frame calls from ServiceSwitcher push_frame
and process_frame.
