v0.3.94

RealtimeSTT 0.3.94

• New Parameters for stop-method of AudioToTextRecorder:

  • backdate_stop_seconds (float, default=0.0):

    • Description: Specifies the number of seconds to backdate the stop time when ending a recording.
    • Usage: When invoking stop() due to a wake word detection or a speaker diarization change event, this parameter compensates for any latency, ensuring that only relevant audio is included in the recording and transcription.

  • backdate_resume_seconds (float, default=0.0):

    • Description: Specifies the number of seconds to backdate the resume time when restarting listening after a recording has stopped.
    • Usage: Typically set to the same value as backdate_stop_seconds, this parameter allows for fine-tuning.

v0.3.93

• fix for stt-server (got broken by webservers dependency upgrade because of an api change)
• added initial_prompt_realtime to AudioToTextRecorder to be able to give different prompts to final and realtime model
• added new parameters to client/server (download root, batch sizes)

v0.3.92

• fixed dependencies (causing "ImportError: cannot import name 'BatchedInferencePipeline' from 'faster_whisper'")

v0.3.91

• upgrade to 0.3.91 since 0.3.9 had issues on PyPi

v0.3.9

RealtimeSTT v0.3.9 Release Notes

🚀 New Features

Batched Transcription

• Added support for batched transcription in both main and real-time models which improves performance and efficiency
• New parameters introduced:
  • batch_size: Controls the batch size for main transcription tasks.
  • realtime_batch_size: Configures batch size for real-time transcription.

This feature is designed to speed up processing. I can't say yet if there may be cases where batching overhead impacts performance negatively. It looked promising for me in initial tests, but I need your feedback! Please report if you get into any issues or notice even slower transcription due to batching.

v0.3.81

RealtimeSTT 0.3.81

Enhanced CLI Interface

• Introduced the -sed command for improved speech end detection
• Added the -l command to set the language
• Implemented the -L command to quickly display a list of all available audio input devices
• Enabled setting the input device index .
• Improved piping support for seamless with > or |

v0.3.7

RealtimeSTT 0.3.7

• fixed a bug to make client terminate gracefully (logged websocket error in debug mode before)
• reworked the CLI interfaces and added shorter commands (for example --writechunks is now -W or --write, for more information please look into the Client Server Readme)

v0.3.6

RealtimeSTT 0.3.6

• more logging for client/server:
  Additional parameters for server:
  • --use_extended_logging, writes extensive log messages for the recording worker, that processes the audio chunks
  • --debug, enables debug logging for detailed server operations
  • --logchunks, enables logging of incoming audio chunks (periods)
  • --writechunks, saves received audio chunks to a WAV file
    Additional parameters for client:
  • --debug, enables debug logging for detailed client operations
  • --writechunks, saves recorded audio chunks to a WAV file
• more logging for AudioToTextRecorder when called with use_extended_logging = True
• new init_realtime_after_seconds parameter for AudioToTextRecorder to finetune the default of 0.2s

v0.3.5

RealtimeSTT 0.3.5

• some upgrades and bugfixes for cli and server (linux support)

v0.3.4

RealtimeSTT 0.3.4

• some upgrades and bugfixes for server
• v0.3.2 yanked