4b704e6d3a
* 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.
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GradiumSTTServicenow takes both anencodingandsample_rateconstructor argument which is assmebled in the class to form theinput_format. PCM accepts8000,16000, and24000Hz sample rates.