RealTimeSTT 0.2.41__tar.gz → 0.3.0__tar.gz
This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.
- {RealtimeSTT-0.2.41 → RealtimeSTT-0.3.0}/PKG-INFO +11 -3
- {RealtimeSTT-0.2.41 → RealtimeSTT-0.3.0}/README.md +10 -2
- {RealtimeSTT-0.2.41 → RealtimeSTT-0.3.0}/RealtimeSTT/audio_recorder.py +756 -285
- RealtimeSTT-0.3.0/RealtimeSTT/outfile.py +2631 -0
- RealtimeSTT-0.3.0/RealtimeSTT/remove_comments.py +78 -0
- {RealtimeSTT-0.2.41 → RealtimeSTT-0.3.0}/RealtimeSTT.egg-info/PKG-INFO +11 -3
- {RealtimeSTT-0.2.41 → RealtimeSTT-0.3.0}/RealtimeSTT.egg-info/SOURCES.txt +2 -0
- {RealtimeSTT-0.2.41 → RealtimeSTT-0.3.0}/setup.py +1 -1
- {RealtimeSTT-0.2.41 → RealtimeSTT-0.3.0}/LICENSE +0 -0
- {RealtimeSTT-0.2.41 → RealtimeSTT-0.3.0}/RealtimeSTT/__init__.py +0 -0
- {RealtimeSTT-0.2.41 → RealtimeSTT-0.3.0}/RealtimeSTT.egg-info/dependency_links.txt +0 -0
- {RealtimeSTT-0.2.41 → RealtimeSTT-0.3.0}/RealtimeSTT.egg-info/requires.txt +0 -0
- {RealtimeSTT-0.2.41 → RealtimeSTT-0.3.0}/RealtimeSTT.egg-info/top_level.txt +0 -0
- {RealtimeSTT-0.2.41 → RealtimeSTT-0.3.0}/setup.cfg +0 -0
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Metadata-Version: 2.1
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Name: RealtimeSTT
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Version: 0.
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Version: 0.3.0
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Summary: A fast Voice Activity Detection and Transcription System
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Home-page: https://github.com/KoljaB/RealTimeSTT
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Author: Kolja Beigel
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- **Voice Assistants**
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- Applications requiring **fast and precise** speech-to-text conversion
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https://github.com/
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https://github.com/user-attachments/assets/797e6552-27cd-41b1-a7f3-e5cbc72094f5
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### Updates
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Latest Version: v0.
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Latest Version: v0.3.0
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See [release history](https://github.com/KoljaB/RealtimeSTT/releases).
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- **debug_mode** (bool, default=False): If set, the system prints additional debug information to the console.
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- **print_transcription_time** (bool, default=False): Logs the processing time of the main model transcription. This can be useful for performance monitoring and debugging.
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- **early_transcription_on_silence** (int, default=0): If set, the system will transcribe audio faster when silence is detected. Transcription will start after the specified milliseconds. Keep this value lower than `post_speech_silence_duration`, ideally around `post_speech_silence_duration` minus the estimated transcription time with the main model. If silence lasts longer than `post_speech_silence_duration`, the recording is stopped, and the transcription is submitted. If voice activity resumes within this period, the transcription is discarded. This results in faster final transcriptions at the cost of additional GPU load due to some unnecessary final transcriptions.
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- **allowed_latency_limit** (int, default=100): Specifies the maximum number of unprocessed chunks in the queue before discarding chunks. This helps prevent the system from being overwhelmed and losing responsiveness in real-time applications.
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- **no_log_file** (bool, default=False): If set, the system will skip writing the debug log file, reducing disk I/O. Useful if logging to a file is not needed and performance is a priority.
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#### Real-time Transcription Parameters
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> **Note**: *When enabling realtime description a GPU installation is strongly advised. Using realtime transcription may create high GPU loads.*
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- **Voice Assistants**
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- Applications requiring **fast and precise** speech-to-text conversion
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https://github.com/
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https://github.com/user-attachments/assets/797e6552-27cd-41b1-a7f3-e5cbc72094f5
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### Updates
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Latest Version: v0.
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Latest Version: v0.3.0
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See [release history](https://github.com/KoljaB/RealtimeSTT/releases).
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- **debug_mode** (bool, default=False): If set, the system prints additional debug information to the console.
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- **print_transcription_time** (bool, default=False): Logs the processing time of the main model transcription. This can be useful for performance monitoring and debugging.
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- **early_transcription_on_silence** (int, default=0): If set, the system will transcribe audio faster when silence is detected. Transcription will start after the specified milliseconds. Keep this value lower than `post_speech_silence_duration`, ideally around `post_speech_silence_duration` minus the estimated transcription time with the main model. If silence lasts longer than `post_speech_silence_duration`, the recording is stopped, and the transcription is submitted. If voice activity resumes within this period, the transcription is discarded. This results in faster final transcriptions at the cost of additional GPU load due to some unnecessary final transcriptions.
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- **allowed_latency_limit** (int, default=100): Specifies the maximum number of unprocessed chunks in the queue before discarding chunks. This helps prevent the system from being overwhelmed and losing responsiveness in real-time applications.
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- **no_log_file** (bool, default=False): If set, the system will skip writing the debug log file, reducing disk I/O. Useful if logging to a file is not needed and performance is a priority.
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#### Real-time Transcription Parameters
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> **Note**: *When enabling realtime description a GPU installation is strongly advised. Using realtime transcription may create high GPU loads.*
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