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.
363 B
363 B
- Improved
GradiumSTTServicetranscription accuracy by reworking how text fragments are accumulated and finalized. Previously, trailing words could be dropped when the server'sflushedresponse arrived before all text tokens were delivered. The service now uses a short aggregation delay after flush to capture trailing tokens, producing complete utterances.