whisperpipe 0.1.0__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.
- whisperpipe-0.1.0/LICENSE +21 -0
- whisperpipe-0.1.0/PKG-INFO +229 -0
- whisperpipe-0.1.0/README.md +190 -0
- whisperpipe-0.1.0/pyproject.toml +67 -0
- whisperpipe-0.1.0/whisperpipe/__init__.py +14 -0
- whisperpipe-0.1.0/whisperpipe/core.py +1627 -0
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MIT License
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Copyright (c) 2024 Erfan Ramezani
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Permission is hereby granted, free of charge, to any person obtaining a copy
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of this software and associated documentation files (the "Software"), to deal
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in the Software without restriction, including without limitation the rights
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to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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copies of the Software, and to permit persons to whom the Software is
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furnished to do so, subject to the following conditions:
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The above copyright notice and this permission notice shall be included in all
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copies or substantial portions of the Software.
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THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
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SOFTWARE.
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Metadata-Version: 2.4
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Name: whisperpipe
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Version: 0.1.0
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Summary: Real-time speech-to-text streaming with OpenAI Whisper
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License: MIT
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License-File: LICENSE
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Keywords: whisper,openai,speech-to-text,stt,asr,automatic speech recognition,real-time,realtime,audio,microphone,transcription,streaming,live transcription
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Author: Erfan Ramezani
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Author-email: erfanramezany245@gmail.com
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Requires-Python: >=3.9,<3.13
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Classifier: Development Status :: 3 - Alpha
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Classifier: Intended Audience :: End Users/Desktop
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Classifier: Intended Audience :: Developers
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Classifier: License :: OSI Approved :: MIT License
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Classifier: Programming Language :: Python :: 3
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Classifier: Programming Language :: Python :: 3.9
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Classifier: Programming Language :: Python :: 3.10
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Classifier: Programming Language :: Python :: 3.11
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Classifier: Programming Language :: Python :: 3.12
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Classifier: Operating System :: POSIX :: Linux
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Classifier: Operating System :: Microsoft :: Windows
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Classifier: Operating System :: MacOS :: MacOS X
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Classifier: Environment :: Console
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Classifier: Topic :: Multimedia :: Sound/Audio :: Speech
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Classifier: Topic :: Multimedia :: Sound/Audio :: Capture/Recording
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Classifier: Topic :: Scientific/Engineering :: Artificial Intelligence
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Classifier: Topic :: Software Development :: Libraries :: Python Modules
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Requires-Dist: openai-whisper (>=20240930)
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Requires-Dist: pyaudio (>=0.2.14)
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Requires-Dist: pynput (>=1.7.7)
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Requires-Dist: sounddevice (>=0.5.2)
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Project-URL: Bug Tracker, https://github.com/Erfan-ram/whisperpipe/issues
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Project-URL: Changelog, https://github.com/Erfan-ram/whisperpipe/releases
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Project-URL: Documentation, https://github.com/Erfan-ram/whisperpipe/blob/main/README.md
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Project-URL: Homepage, https://github.com/Erfan-ram/whisperpipe
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Project-URL: Repository, https://github.com/Erfan-ram/whisperpipe
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Project-URL: Source, https://github.com/Erfan-ram/whisperpipe
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Description-Content-Type: text/markdown
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# whisperpipe
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Real-time speech-to-text streaming with OpenAI Whisper
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## Description
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whisperpipe is a powerful, easy-to-use Python package for real-time, offline audio transcription using OpenAI's Whisper model. It runs locally, making it a free and private solution for continuous speech-to-text applications. It provides seamless integration with callback functions for LLM processing and supports pause/resume functionality for interactive applications.
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## Why whisperpipe?
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In a world where most ASR (Automatic Speech Recognition) services are cloud-based, whisperpipe offers a refreshing alternative by harnessing the power of OpenAI's Whisper model to run directly on your local machine. This approach provides several key advantages:
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- **Complete Privacy**: Since all transcription is done locally, your voice data never leaves your computer. This is crucial for applications that handle sensitive or private conversations.
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- **Zero Cost**: Say goodbye to recurring subscription fees and per-minute charges. whisperpipe is free to use, making it an economical choice for both hobbyists and commercial projects.
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- **No Internet Required**: Whether you're on a plane, in a remote location, or simply have an unstable internet connection, whisperpipe works flawlessly offline.
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- **Real-time Performance**: Designed for continuous, real-time transcription, whisperpipe is ideal for live applications such as voice-controlled assistants, dictation software, and more.
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- **Unleash the Power of Whisper**: By running the Whisper model locally, you have full control over the transcription process, from model selection to performance tuning.
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whisperpipe empowers you to build powerful, private, and cost-effective voice applications with ease.
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## Features
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- **Real-time audio transcription** using OpenAI Whisper
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- **Callback system** for custom processing (LLM integration, etc.)
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- **Pause/Resume functionality** for interactive applications
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- **Multiple language support**
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- **Configurable processing parameters**
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- **Thread-safe operation**
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- **Easy installation and usage**
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## Installation
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### From PyPI
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```bash
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pip install whisperpipe
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```
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### From GitHub
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```bash
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pip install git+https://github.com/Erfan-ram/whisperpipe.git
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```
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## Quick Start
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```python
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from whisperpipe import pipeStream
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# Basic usage
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transcriber = pipeStream(
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model_name="base",
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language="en",
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finalization_delay=10.0,
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processing_interval=1.0
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)
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# Start streaming
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transcriber.start_streaming()
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```
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## Usage Examples
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### Basic Transcription
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```python
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from whisperpipe import pipeStream
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# Create transcriber instance
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transcriber = pipeStream(
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model_name="base",
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language="en",
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finalization_delay=10.0,
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processing_interval=1.0
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)
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# Start transcription
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transcriber.start_streaming()
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# The transcribed text will be printed to console
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# Press Ctrl+C to stop
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```
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### With Custom Callback (LLM Integration)
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```python
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from whisperpipe import pipeStream
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def llm_processor(text):
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"""Custom function to process transcribed text"""
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print(f"Processing: {text}")
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# Your LLM integration here
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# e.g., send to OpenAI, Claude, local model, etc.
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response = your_llm_api.chat(text)
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print(f"Response: {response}")
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return response
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# Create transcriber with callback
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transcriber = pipeStream(
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model_name="base",
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language="en",
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finalization_delay=10.0,
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processing_interval=1.0
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)
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# Register callback
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transcriber.set_def_callback(llm_processor)
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# Start streaming with LLM integration
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transcriber.start_streaming()
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```
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### Interactive Mode with Pause/Resume
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```python
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from whisperpipe import pipeStream
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import time
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def interactive_processor(text):
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"""Process text and pause for response"""
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# Pause transcriber while processing
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transcriber.pause_streaming()
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print(f"User said: {text}")
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# Process with your system
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response = process_with_llm(text)
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# Speak or display response
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print(f"Assistant: {response}")
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# Resume for next input
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transcriber.resume_streaming()
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transcriber = pipeStream()
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transcriber.set_def_callback(interactive_processor)
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transcriber.start_streaming()
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```
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## API Reference
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### Constructor Parameters
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- `model_name` (str): Whisper model name ("tiny", "base", "small", "medium", "large"). Default: "base"
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- `language` (str): Language code for transcription ("en", "es", "fr", etc.). Default: "en"
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- `finalization_delay` (float): Wait time in seconds before finalizing transcription. Default: 10.0
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- `processing_interval` (float): Interval in seconds between processing cycles. Default: 1.0
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- `buffer_duration_seconds` (float): Time window in seconds to hold audio for processing. Default: 5.0
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- `debug_mode` (bool): Enable debug mode for detailed logging. Default: True
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### Methods
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#### Core Methods
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- `start_streaming()`: Start audio capture and transcription
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- `stop_streaming()`: Stop audio capture and transcription
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#### Callback System
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- `set_def_callback(callback_function)`: Register a callback function for processing transcribed text
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- `set_def_callback(None)`: Clear the callback (use default behavior)
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#### Pause/Resume Control
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- `pause_streaming()`: Pause audio processing temporarily
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- `resume_streaming()`: Resume audio processing
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- `is_paused()`: Check if transcriber is paused
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- `is_running()`: Check if transcriber is running
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## Requirements
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- Python 3.8+
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- PyAudio
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- OpenAI Whisper
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- PyTorch
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- NumPy
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- pynput
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## License
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MIT License
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## Author
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Erfan Ramezani - erfanramezany245@gmail.com
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## Contributing
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Contributions are welcome! Please feel free to submit a Pull Request.
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## Support
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For issues and questions, please use the [GitHub Issues](https://github.com/Erfan-ram/whisperpipe/issues) page.
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@@ -0,0 +1,190 @@
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# whisperpipe
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Real-time speech-to-text streaming with OpenAI Whisper
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## Description
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whisperpipe is a powerful, easy-to-use Python package for real-time, offline audio transcription using OpenAI's Whisper model. It runs locally, making it a free and private solution for continuous speech-to-text applications. It provides seamless integration with callback functions for LLM processing and supports pause/resume functionality for interactive applications.
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## Why whisperpipe?
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In a world where most ASR (Automatic Speech Recognition) services are cloud-based, whisperpipe offers a refreshing alternative by harnessing the power of OpenAI's Whisper model to run directly on your local machine. This approach provides several key advantages:
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- **Complete Privacy**: Since all transcription is done locally, your voice data never leaves your computer. This is crucial for applications that handle sensitive or private conversations.
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14
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- **Zero Cost**: Say goodbye to recurring subscription fees and per-minute charges. whisperpipe is free to use, making it an economical choice for both hobbyists and commercial projects.
|
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15
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+
- **No Internet Required**: Whether you're on a plane, in a remote location, or simply have an unstable internet connection, whisperpipe works flawlessly offline.
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16
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- **Real-time Performance**: Designed for continuous, real-time transcription, whisperpipe is ideal for live applications such as voice-controlled assistants, dictation software, and more.
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- **Unleash the Power of Whisper**: By running the Whisper model locally, you have full control over the transcription process, from model selection to performance tuning.
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whisperpipe empowers you to build powerful, private, and cost-effective voice applications with ease.
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## Features
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- **Real-time audio transcription** using OpenAI Whisper
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- **Callback system** for custom processing (LLM integration, etc.)
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- **Pause/Resume functionality** for interactive applications
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- **Multiple language support**
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- **Configurable processing parameters**
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- **Thread-safe operation**
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- **Easy installation and usage**
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## Installation
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### From PyPI
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```bash
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pip install whisperpipe
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```
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### From GitHub
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```bash
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pip install git+https://github.com/Erfan-ram/whisperpipe.git
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```
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## Quick Start
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```python
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from whisperpipe import pipeStream
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# Basic usage
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transcriber = pipeStream(
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model_name="base",
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language="en",
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finalization_delay=10.0,
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processing_interval=1.0
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)
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# Start streaming
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transcriber.start_streaming()
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```
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## Usage Examples
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### Basic Transcription
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```python
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from whisperpipe import pipeStream
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# Create transcriber instance
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transcriber = pipeStream(
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model_name="base",
|
|
72
|
+
language="en",
|
|
73
|
+
finalization_delay=10.0,
|
|
74
|
+
processing_interval=1.0
|
|
75
|
+
)
|
|
76
|
+
|
|
77
|
+
# Start transcription
|
|
78
|
+
transcriber.start_streaming()
|
|
79
|
+
|
|
80
|
+
# The transcribed text will be printed to console
|
|
81
|
+
# Press Ctrl+C to stop
|
|
82
|
+
```
|
|
83
|
+
|
|
84
|
+
### With Custom Callback (LLM Integration)
|
|
85
|
+
|
|
86
|
+
```python
|
|
87
|
+
from whisperpipe import pipeStream
|
|
88
|
+
|
|
89
|
+
def llm_processor(text):
|
|
90
|
+
"""Custom function to process transcribed text"""
|
|
91
|
+
print(f"Processing: {text}")
|
|
92
|
+
# Your LLM integration here
|
|
93
|
+
# e.g., send to OpenAI, Claude, local model, etc.
|
|
94
|
+
response = your_llm_api.chat(text)
|
|
95
|
+
print(f"Response: {response}")
|
|
96
|
+
return response
|
|
97
|
+
|
|
98
|
+
# Create transcriber with callback
|
|
99
|
+
transcriber = pipeStream(
|
|
100
|
+
model_name="base",
|
|
101
|
+
language="en",
|
|
102
|
+
finalization_delay=10.0,
|
|
103
|
+
processing_interval=1.0
|
|
104
|
+
)
|
|
105
|
+
|
|
106
|
+
# Register callback
|
|
107
|
+
transcriber.set_def_callback(llm_processor)
|
|
108
|
+
|
|
109
|
+
# Start streaming with LLM integration
|
|
110
|
+
transcriber.start_streaming()
|
|
111
|
+
```
|
|
112
|
+
|
|
113
|
+
### Interactive Mode with Pause/Resume
|
|
114
|
+
|
|
115
|
+
```python
|
|
116
|
+
from whisperpipe import pipeStream
|
|
117
|
+
import time
|
|
118
|
+
|
|
119
|
+
def interactive_processor(text):
|
|
120
|
+
"""Process text and pause for response"""
|
|
121
|
+
# Pause transcriber while processing
|
|
122
|
+
transcriber.pause_streaming()
|
|
123
|
+
|
|
124
|
+
print(f"User said: {text}")
|
|
125
|
+
|
|
126
|
+
# Process with your system
|
|
127
|
+
response = process_with_llm(text)
|
|
128
|
+
|
|
129
|
+
# Speak or display response
|
|
130
|
+
print(f"Assistant: {response}")
|
|
131
|
+
|
|
132
|
+
# Resume for next input
|
|
133
|
+
transcriber.resume_streaming()
|
|
134
|
+
|
|
135
|
+
transcriber = pipeStream()
|
|
136
|
+
transcriber.set_def_callback(interactive_processor)
|
|
137
|
+
transcriber.start_streaming()
|
|
138
|
+
```
|
|
139
|
+
|
|
140
|
+
## API Reference
|
|
141
|
+
|
|
142
|
+
### Constructor Parameters
|
|
143
|
+
|
|
144
|
+
- `model_name` (str): Whisper model name ("tiny", "base", "small", "medium", "large"). Default: "base"
|
|
145
|
+
- `language` (str): Language code for transcription ("en", "es", "fr", etc.). Default: "en"
|
|
146
|
+
- `finalization_delay` (float): Wait time in seconds before finalizing transcription. Default: 10.0
|
|
147
|
+
- `processing_interval` (float): Interval in seconds between processing cycles. Default: 1.0
|
|
148
|
+
- `buffer_duration_seconds` (float): Time window in seconds to hold audio for processing. Default: 5.0
|
|
149
|
+
- `debug_mode` (bool): Enable debug mode for detailed logging. Default: True
|
|
150
|
+
|
|
151
|
+
### Methods
|
|
152
|
+
|
|
153
|
+
#### Core Methods
|
|
154
|
+
- `start_streaming()`: Start audio capture and transcription
|
|
155
|
+
- `stop_streaming()`: Stop audio capture and transcription
|
|
156
|
+
|
|
157
|
+
#### Callback System
|
|
158
|
+
- `set_def_callback(callback_function)`: Register a callback function for processing transcribed text
|
|
159
|
+
- `set_def_callback(None)`: Clear the callback (use default behavior)
|
|
160
|
+
|
|
161
|
+
#### Pause/Resume Control
|
|
162
|
+
- `pause_streaming()`: Pause audio processing temporarily
|
|
163
|
+
- `resume_streaming()`: Resume audio processing
|
|
164
|
+
- `is_paused()`: Check if transcriber is paused
|
|
165
|
+
- `is_running()`: Check if transcriber is running
|
|
166
|
+
|
|
167
|
+
## Requirements
|
|
168
|
+
|
|
169
|
+
- Python 3.8+
|
|
170
|
+
- PyAudio
|
|
171
|
+
- OpenAI Whisper
|
|
172
|
+
- PyTorch
|
|
173
|
+
- NumPy
|
|
174
|
+
- pynput
|
|
175
|
+
|
|
176
|
+
## License
|
|
177
|
+
|
|
178
|
+
MIT License
|
|
179
|
+
|
|
180
|
+
## Author
|
|
181
|
+
|
|
182
|
+
Erfan Ramezani - erfanramezany245@gmail.com
|
|
183
|
+
|
|
184
|
+
## Contributing
|
|
185
|
+
|
|
186
|
+
Contributions are welcome! Please feel free to submit a Pull Request.
|
|
187
|
+
|
|
188
|
+
## Support
|
|
189
|
+
|
|
190
|
+
For issues and questions, please use the [GitHub Issues](https://github.com/Erfan-ram/whisperpipe/issues) page.
|
|
@@ -0,0 +1,67 @@
|
|
|
1
|
+
[project]
|
|
2
|
+
name = "whisperpipe"
|
|
3
|
+
version = "0.1.0"
|
|
4
|
+
description = "Real-time speech-to-text streaming with OpenAI Whisper"
|
|
5
|
+
authors = [
|
|
6
|
+
{name = "Erfan Ramezani", email = "erfanramezany245@gmail.com"}
|
|
7
|
+
]
|
|
8
|
+
license = {text = "MIT"}
|
|
9
|
+
readme = "README.md"
|
|
10
|
+
keywords = [
|
|
11
|
+
"whisper", "openai", "speech-to-text", "stt", "asr",
|
|
12
|
+
"automatic speech recognition", "real-time", "realtime",
|
|
13
|
+
"audio", "microphone", "transcription", "streaming", "live transcription"
|
|
14
|
+
]
|
|
15
|
+
classifiers = [
|
|
16
|
+
"Development Status :: 3 - Alpha",
|
|
17
|
+
"Intended Audience :: End Users/Desktop",
|
|
18
|
+
"Intended Audience :: Developers",
|
|
19
|
+
"License :: OSI Approved :: MIT License",
|
|
20
|
+
"Programming Language :: Python :: 3",
|
|
21
|
+
"Programming Language :: Python :: 3.9",
|
|
22
|
+
"Programming Language :: Python :: 3.10",
|
|
23
|
+
"Programming Language :: Python :: 3.11",
|
|
24
|
+
"Programming Language :: Python :: 3.12",
|
|
25
|
+
"Operating System :: POSIX :: Linux",
|
|
26
|
+
"Operating System :: Microsoft :: Windows",
|
|
27
|
+
"Operating System :: MacOS :: MacOS X",
|
|
28
|
+
"Environment :: Console",
|
|
29
|
+
"Topic :: Multimedia :: Sound/Audio :: Speech",
|
|
30
|
+
"Topic :: Multimedia :: Sound/Audio :: Capture/Recording",
|
|
31
|
+
"Topic :: Scientific/Engineering :: Artificial Intelligence",
|
|
32
|
+
"Topic :: Software Development :: Libraries :: Python Modules"
|
|
33
|
+
]
|
|
34
|
+
requires-python = ">=3.9,<3.13"
|
|
35
|
+
dependencies = [
|
|
36
|
+
"openai-whisper>=20240930",
|
|
37
|
+
"pyaudio>=0.2.14",
|
|
38
|
+
"pynput>=1.7.7",
|
|
39
|
+
"sounddevice>=0.5.2"
|
|
40
|
+
]
|
|
41
|
+
|
|
42
|
+
[project.urls]
|
|
43
|
+
Homepage = "https://github.com/Erfan-ram/whisperpipe"
|
|
44
|
+
Repository = "https://github.com/Erfan-ram/whisperpipe"
|
|
45
|
+
Documentation = "https://github.com/Erfan-ram/whisperpipe/blob/main/README.md"
|
|
46
|
+
"Bug Tracker" = "https://github.com/Erfan-ram/whisperpipe/issues"
|
|
47
|
+
"Source" = "https://github.com/Erfan-ram/whisperpipe"
|
|
48
|
+
Changelog = "https://github.com/Erfan-ram/whisperpipe/releases"
|
|
49
|
+
|
|
50
|
+
[tool.poetry.group.dev.dependencies]
|
|
51
|
+
pytest = "^7.0"
|
|
52
|
+
black = "^23.0"
|
|
53
|
+
flake8 = "^6.0"
|
|
54
|
+
|
|
55
|
+
[build-system]
|
|
56
|
+
requires = ["poetry-core"]
|
|
57
|
+
build-backend = "poetry.core.masonry.api"
|
|
58
|
+
|
|
59
|
+
[tool.pytest.ini_options]
|
|
60
|
+
testpaths = ["tests"]
|
|
61
|
+
python_files = ["test_*.py", "*_test.py"]
|
|
62
|
+
python_functions = ["test_*"]
|
|
63
|
+
addopts = "-v --tb=short"
|
|
64
|
+
filterwarnings = [
|
|
65
|
+
"ignore::DeprecationWarning",
|
|
66
|
+
"ignore::PendingDeprecationWarning"
|
|
67
|
+
]
|
|
@@ -0,0 +1,14 @@
|
|
|
1
|
+
"""
|
|
2
|
+
whisperpipe - Real-time speech-to-text streaming with OpenAI Whisper
|
|
3
|
+
|
|
4
|
+
A powerful, easy-to-use package for real-time audio transcription using OpenAI's Whisper model.
|
|
5
|
+
Features include callback support for LLM integration and pause/resume functionality.
|
|
6
|
+
"""
|
|
7
|
+
|
|
8
|
+
from .core import pipeStream
|
|
9
|
+
|
|
10
|
+
__version__ = "1.0.0"
|
|
11
|
+
__author__ = "Erfan Ramezani"
|
|
12
|
+
__email__ = "erfanramezany245@gmail.com"
|
|
13
|
+
|
|
14
|
+
__all__ = ["pipeStream"]
|