live-translation 0.1.0__tar.gz

live_translation-0.1.0/LICENSE

@@ -0,0 +1,20 @@
+MIT License
+
+Copyright (c) 2024 Abdullah Hendy
+
+Permission is hereby granted, free of charge, to any person obtaining a copy
+of this software and associated documentation files (the "Software"), to deal
+in the Software without restriction, including without limitation the rights
+to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+copies of the Software, and to permit persons to whom the Software is furnished 
+to do so, subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in all 
+copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR 
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS 
+FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR 
+COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER 
+IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN 
+CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.

live_translation-0.1.0/PKG-INFO

@@ -0,0 +1,296 @@
+Metadata-Version: 2.4
+Name: live-translation
+Version: 0.1.0
+Summary: A real-time translation tool using Whisper & Opus-MT
+Author-email: Abdullah Hendy <abdullah.a.hendy@gmail.com>
+Project-URL: Homepage, https://github.com/AbdullahHendy/live-translation
+Project-URL: Repository, https://github.com/AbdullahHendy/live-translation
+Keywords: translation,transcription,speech-to-text,real-time translation,whisper,marianMT,Opus-MT,live transcription
+Requires-Python: >=3.8
+Description-Content-Type: text/markdown
+License-File: LICENSE
+Requires-Dist: torch
+Requires-Dist: torchaudio
+Requires-Dist: transformers
+Requires-Dist: numpy
+Requires-Dist: pyaudio
+Requires-Dist: websockets
+Requires-Dist: huggingface_hub
+Requires-Dist: sacremoses
+Requires-Dist: faster-whisper
+Requires-Dist: sentencepiece
+Dynamic: license-file
+
+# Real-time Speech-to-Text Translation
+
+This project provides a real-time speech-to-text translation solution. It captures audio from the microphone, processes it, transcribes it into text, and translates it to a target language. It uses the **Silero** model for processing (Voice Activity Detection), **Whisper** model for transcription and **Opus-MT** for translation. The output can be through ***stdout***, a ***JSON file***, or ***websockets***. 
+
+## Features
+
+- Real-time speech capture and processing using **Silero** VAD (Voice Activity Detection)
+- Speech-to-text transcription using the Whisper model
+- Translation of transcriptions from a source language to a target language
+- Multithreaded design for efficient processing
+- Different output modes: stdout, **JSON** file, websocket server
+
+## Prerequisites
+
+Before running the project, you need to install the following system dependencies:
+
+- **PortAudio** (for audio input handling)
+- **FFmpeg** (for audio and video processing)
+    - On Ubuntu/Debian-based systems, you can install it with:
+      ```bash
+      sudo apt-get install portaudio19-dev ffmpeg
+      ```
+
+## Installation
+
+**(RECOMMENDED)**: install this package inside a virtual environment to avoid dependency conflicts.
+```bash
+python -m venv .venv
+source .venv/bin/activate
+```
+
+**Install** the package from PyPI:
+```bash
+pip install live-translation
+```
+
+**Verify** the installation:
+```bash
+python -c "import live_translation; print('live-translation installed successfully!')"
+```
+
+## Usage
+
+> **NOTE**: One can safely ignore the following warning that might appear on **Linux** systems:
+>
+> ALSA lib pcm_dsnoop.c:567:(snd_pcm_dsnoop_open) unable to open slave
+> ALSA lib pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave
+> ALSA lib pcm.c:2722:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear
+> ALSA lib pcm.c:2722:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe
+> ALSA lib pcm.c:2722:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side
+> ALSA lib pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave
+> Cannot connect to server socket err = No such file or directory
+> Cannot connect to server request channel
+> jack server is not running or cannot be started
+> JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock
+> JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock
+>
+
+### CLI 
+live-translation can be run directly from the command line:
+```bash
+live-translate [OPTIONS]
+```
+
+**[OPTIONS]**
+```bash
+usage: live-translate [-h] [--silence_threshold SILENCE_THRESHOLD] [--vad_aggressiveness {0,1,2,3,4,5,6,7,8,9}] [--max_buffer_duration {5,6,7,8,9,10}] [--device {cpu,cuda}] [--whisper_model {tiny,base,small,medium,large,large-v2}]
+                      [--trans_model {Helsinki-NLP/opus-mt,Helsinki-NLP/opus-mt-tc-big}] [--src_lang SRC_LANG] [--tgt_lang TGT_LANG] [--output {print,file,websocket}] [--ws_port WS_PORT] [--transcribe_only]
+
+Live Translation Pipeline - Configure runtime settings.
+
+options:
+  -h, --help            show this help message and exit
+  --silence_threshold SILENCE_THRESHOLD
+                        Number of consecutive 32ms silent chunks to detect SILENCE.
+                        SILENCE clears the audio buffer for transcription/translation.
+                        NOTE: Minimum value is 16.
+                        Default is 65 (~ 2s).
+  --vad_aggressiveness {0,1,2,3,4,5,6,7,8,9}
+                        Voice Activity Detection (VAD) aggressiveness level (0-9).
+                        Higher values mean VAD has to be more confident to detect speech vs silence.
+                        Default is 8.
+  --max_buffer_duration {5,6,7,8,9,10}
+                        Max audio buffer duration in seconds before trimming it.
+                        Default is 7 seconds.
+  --device {cpu,cuda}   Device for processing ('cpu', 'cuda').
+                        Default is 'cpu'.
+  --whisper_model {tiny,base,small,medium,large,large-v2}
+                        Whisper model size ('tiny', 'base', 'small', 'medium', 'large', 'large-v2').
+                        Default is 'base'.
+  --trans_model {Helsinki-NLP/opus-mt,Helsinki-NLP/opus-mt-tc-big}
+                        Translation model ('Helsinki-NLP/opus-mt', 'Helsinki-NLP/opus-mt-tc-big'). 
+                        NOTE: Don't include source and target languages here.
+                        Default is 'Helsinki-NLP/opus-mt'.
+  --src_lang SRC_LANG   Source/Input language for transcription (e.g., 'en', 'fr').
+                        Default is 'en'.
+  --tgt_lang TGT_LANG   Target language for translation (e.g., 'es', 'de').
+                        Default is 'es'.
+  --output {print,file,websocket}
+                        Output method ('print', 'file', 'websocket').
+                          - 'print': Prints transcriptions and translations to stdout.
+                          - 'file': Saves structured JSON data (see below) in ./transcripts/transcriptions.json.
+                          - 'websocket': Sends structured JSON data (see below) over WebSocket.
+                        JSON format for 'file' and 'websocket':
+                        {
+                            "timestamp": "2025-03-06T12:34:56.789Z",
+                            "transcription": "Hello world",
+                            "translation": "Hola mundo"
+                        }.
+                        Default is 'print'.
+  --ws_port WS_PORT     WebSocket port for sending transcriptions.
+                        Required if --output is 'websocket'.
+  --transcribe_only     Transcribe only mode. No translations are performed.
+```
+
+- in case of **websockets**, one can connect to the server using **curl**, **wscat**, etc.. 
+  ```bash
+  curl --include --no-buffer ws://localhost:<PORT_NUM>
+  ```
+  ```bash
+  wscat -c ws://localhost:<PORT_NUM>
+  ```
+
+### API
+You can also import and use live_translation directly in your Python code.
+The following is a ***simple*** example of running *live_translation* in a server/client fashion.
+For more detailed examples see [examples/](/examples/).
+
+- **Server**
+  ```python
+  from live_translation.config import Config
+  from live_translation.app import LiveTranslationApp
+
+  # Initialize the app with custom configuration
+  # Configure to be CPU-based and output to WebSocket
+  config = Config(
+    device="cpu",
+    output="websocket",
+    ws_port=8765,
+  )
+
+  # Create and start the Live Translation App
+  app = LiveTranslationApp(config)
+  app.run()
+  ```
+
+- **Client**
+  ```python
+  import asyncio
+  import websockets
+  import json
+
+  async def listen():
+      uri = "ws://localhost:8765"
+      async with websockets.connect(uri) as websocket:
+          print("🔌 Connected to Live Translation WebSocket server.")
+
+          try:
+              while True:
+                  message = await websocket.recv()
+                  data = json.loads(message)
+
+                  print(f"⏳ Timestamp: {data['timestamp']}")
+                  print(f"📝 Transcription: {data['transcription']}")
+                  print(f"🌍 Translation: {data['translation']}\n")
+
+          except websockets.exceptions.ConnectionClosed:
+              print("WebSocket connection closed.")
+
+  asyncio.run(listen())
+  ```
+
+## Development
+
+To contribute or modify this project, these steps might be helpful:
+
+**Clone** the repository:
+```bash
+git clone git@github.com:AbdullahHendy/live-translation.git
+cd live-translation
+```
+
+**Ceate** a virtual environment:
+```bash
+python -m venv .venv
+source .venv/bin/activate 
+```
+
+**Install** Dependencies:
+```bash
+pip install --upgrade pip
+pip install -r requirements.txt
+```
+
+**Test** the package:
+```bash
+make test
+```
+
+**Build** the package:
+```bash
+make build
+```
+> **NOTE**: Building generates a ***.whl*** file that can be ***pip*** insalled in a new environment for testing
+
+**If needed**, run the program within the virtual environment:
+```bash
+python -m live_translation.cli [OPTIONS]
+```
+
+## Tested Environment
+
+This project was tested and developed on the following system configuration:
+
+- **Architecture**: x86_64 (64-bit)
+- **Operating System**: Ubuntu 24.10 (Oracular Oriole)
+- **Kernel Version**: 6.11.0-18-generic
+- **Python Version**: 3.12.7
+- **Processor**: 13th Gen Intel(R) Core(TM) i9-13900HX
+- **GPU**: GeForce RTX 4070 Max-Q / Mobile [^1]
+- **RAM**: 16GB DDR5
+- **Dependencies**: All required dependencies are listed in `requirements.txt` and [Prerequisites](#prerequisites)
+
+[^1]: CUDA not utilized, as the `DEVICE` configuration is set to `"cpu"`. Additional Nvidia drivers, CUDA, cuDNN installation needed if option `"cuda"` were to be used.
+
+## Improvements
+
+- **Block Diagram**: Include a block diagram to visually represent the flow and architecture of the system, making it easier to understand the overall design.
+- **Better Error Handling**: Improve error handling across various components (audio, transcription, translation) to ensure the system is robust and can handle unexpected scenarios gracefully.
+- **Performance Optimization**: Investigate performance bottlenecks including checking sleep durations and optimizing concurrency management to minimize lag.
+- **Concurrency Design Check**: Review and optimize the threading design to ensure thread safety and prevent issues like race conditions or deadlocks, etc., revisit the current design of ***AudioRecorder*** being a thread while ***AudioProcessor***, ***Transcriber***, and ***Translator*** being processes.
+- **Logging**: Integrate detailed logging to track system activity, errors, and performance metrics using a more formal logging framework.
+
+## Citations
+ ```bibtex
+  @article{Whisper,
+    title = {Robust Speech Recognition via Large-Scale Weak Supervision},
+    url = {https://arxiv.org/abs/2212.04356},
+    author = {Radford, Alec and Kim, Jong Wook and Xu, Tao and Brockman, Greg and McLeavey, Christine and Sutskever, Ilya},
+    publisher = {arXiv},
+    year = {2022}
+  }
+
+  @misc{Silero VAD,
+    author = {Silero Team},
+    title = {Silero VAD: pre-trained enterprise-grade Voice Activity Detector (VAD), Number Detector and Language Classifier},
+    year = {2021},
+    publisher = {GitHub},
+    journal = {GitHub repository},
+    howpublished = {\url{https://github.com/snakers4/silero-vad}},
+    email = {hello@silero.ai}
+  }
+
+  @article{tiedemann2023democratizing,
+    title={Democratizing neural machine translation with {OPUS-MT}},
+    author={Tiedemann, J{\"o}rg and Aulamo, Mikko and Bakshandaeva, Daria and Boggia, Michele and Gr{\"o}nroos, Stig-Arne and Nieminen, Tommi and Raganato, Alessandro and Scherrer, Yves and Vazquez, Raul and Virpioja, Sami},
+    journal={Language Resources and Evaluation},
+    number={58},
+    pages={713--755},
+    year={2023},
+    publisher={Springer Nature},
+    issn={1574-0218},
+    doi={10.1007/s10579-023-09704-w}
+  }
+
+  @InProceedings{TiedemannThottingal:EAMT2020,
+    author = {J{\"o}rg Tiedemann and Santhosh Thottingal},
+    title = {{OPUS-MT} — {B}uilding open translation services for the {W}orld},
+    booktitle = {Proceedings of the 22nd Annual Conference of the European Association for Machine Translation (EAMT)},
+    year = {2020},
+    address = {Lisbon, Portugal}
+  }
+```

live_translation-0.1.0/README.md

@@ -0,0 +1,273 @@
+# Real-time Speech-to-Text Translation
+
+This project provides a real-time speech-to-text translation solution. It captures audio from the microphone, processes it, transcribes it into text, and translates it to a target language. It uses the **Silero** model for processing (Voice Activity Detection), **Whisper** model for transcription and **Opus-MT** for translation. The output can be through ***stdout***, a ***JSON file***, or ***websockets***. 
+
+## Features
+
+- Real-time speech capture and processing using **Silero** VAD (Voice Activity Detection)
+- Speech-to-text transcription using the Whisper model
+- Translation of transcriptions from a source language to a target language
+- Multithreaded design for efficient processing
+- Different output modes: stdout, **JSON** file, websocket server
+
+## Prerequisites
+
+Before running the project, you need to install the following system dependencies:
+
+- **PortAudio** (for audio input handling)
+- **FFmpeg** (for audio and video processing)
+    - On Ubuntu/Debian-based systems, you can install it with:
+      ```bash
+      sudo apt-get install portaudio19-dev ffmpeg
+      ```
+
+## Installation
+
+**(RECOMMENDED)**: install this package inside a virtual environment to avoid dependency conflicts.
+```bash
+python -m venv .venv
+source .venv/bin/activate
+```
+
+**Install** the package from PyPI:
+```bash
+pip install live-translation
+```
+
+**Verify** the installation:
+```bash
+python -c "import live_translation; print('live-translation installed successfully!')"
+```
+
+## Usage
+
+> **NOTE**: One can safely ignore the following warning that might appear on **Linux** systems:
+>
+> ALSA lib pcm_dsnoop.c:567:(snd_pcm_dsnoop_open) unable to open slave
+> ALSA lib pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave
+> ALSA lib pcm.c:2722:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear
+> ALSA lib pcm.c:2722:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe
+> ALSA lib pcm.c:2722:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side
+> ALSA lib pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave
+> Cannot connect to server socket err = No such file or directory
+> Cannot connect to server request channel
+> jack server is not running or cannot be started
+> JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock
+> JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock
+>
+
+### CLI 
+live-translation can be run directly from the command line:
+```bash
+live-translate [OPTIONS]
+```
+
+**[OPTIONS]**
+```bash
+usage: live-translate [-h] [--silence_threshold SILENCE_THRESHOLD] [--vad_aggressiveness {0,1,2,3,4,5,6,7,8,9}] [--max_buffer_duration {5,6,7,8,9,10}] [--device {cpu,cuda}] [--whisper_model {tiny,base,small,medium,large,large-v2}]
+                      [--trans_model {Helsinki-NLP/opus-mt,Helsinki-NLP/opus-mt-tc-big}] [--src_lang SRC_LANG] [--tgt_lang TGT_LANG] [--output {print,file,websocket}] [--ws_port WS_PORT] [--transcribe_only]
+
+Live Translation Pipeline - Configure runtime settings.
+
+options:
+  -h, --help            show this help message and exit
+  --silence_threshold SILENCE_THRESHOLD
+                        Number of consecutive 32ms silent chunks to detect SILENCE.
+                        SILENCE clears the audio buffer for transcription/translation.
+                        NOTE: Minimum value is 16.
+                        Default is 65 (~ 2s).
+  --vad_aggressiveness {0,1,2,3,4,5,6,7,8,9}
+                        Voice Activity Detection (VAD) aggressiveness level (0-9).
+                        Higher values mean VAD has to be more confident to detect speech vs silence.
+                        Default is 8.
+  --max_buffer_duration {5,6,7,8,9,10}
+                        Max audio buffer duration in seconds before trimming it.
+                        Default is 7 seconds.
+  --device {cpu,cuda}   Device for processing ('cpu', 'cuda').
+                        Default is 'cpu'.
+  --whisper_model {tiny,base,small,medium,large,large-v2}
+                        Whisper model size ('tiny', 'base', 'small', 'medium', 'large', 'large-v2').
+                        Default is 'base'.
+  --trans_model {Helsinki-NLP/opus-mt,Helsinki-NLP/opus-mt-tc-big}
+                        Translation model ('Helsinki-NLP/opus-mt', 'Helsinki-NLP/opus-mt-tc-big'). 
+                        NOTE: Don't include source and target languages here.
+                        Default is 'Helsinki-NLP/opus-mt'.
+  --src_lang SRC_LANG   Source/Input language for transcription (e.g., 'en', 'fr').
+                        Default is 'en'.
+  --tgt_lang TGT_LANG   Target language for translation (e.g., 'es', 'de').
+                        Default is 'es'.
+  --output {print,file,websocket}
+                        Output method ('print', 'file', 'websocket').
+                          - 'print': Prints transcriptions and translations to stdout.
+                          - 'file': Saves structured JSON data (see below) in ./transcripts/transcriptions.json.
+                          - 'websocket': Sends structured JSON data (see below) over WebSocket.
+                        JSON format for 'file' and 'websocket':
+                        {
+                            "timestamp": "2025-03-06T12:34:56.789Z",
+                            "transcription": "Hello world",
+                            "translation": "Hola mundo"
+                        }.
+                        Default is 'print'.
+  --ws_port WS_PORT     WebSocket port for sending transcriptions.
+                        Required if --output is 'websocket'.
+  --transcribe_only     Transcribe only mode. No translations are performed.
+```
+
+- in case of **websockets**, one can connect to the server using **curl**, **wscat**, etc.. 
+  ```bash
+  curl --include --no-buffer ws://localhost:<PORT_NUM>
+  ```
+  ```bash
+  wscat -c ws://localhost:<PORT_NUM>
+  ```
+
+### API
+You can also import and use live_translation directly in your Python code.
+The following is a ***simple*** example of running *live_translation* in a server/client fashion.
+For more detailed examples see [examples/](/examples/).
+
+- **Server**
+  ```python
+  from live_translation.config import Config
+  from live_translation.app import LiveTranslationApp
+
+  # Initialize the app with custom configuration
+  # Configure to be CPU-based and output to WebSocket
+  config = Config(
+    device="cpu",
+    output="websocket",
+    ws_port=8765,
+  )
+
+  # Create and start the Live Translation App
+  app = LiveTranslationApp(config)
+  app.run()
+  ```
+
+- **Client**
+  ```python
+  import asyncio
+  import websockets
+  import json
+
+  async def listen():
+      uri = "ws://localhost:8765"
+      async with websockets.connect(uri) as websocket:
+          print("🔌 Connected to Live Translation WebSocket server.")
+
+          try:
+              while True:
+                  message = await websocket.recv()
+                  data = json.loads(message)
+
+                  print(f"⏳ Timestamp: {data['timestamp']}")
+                  print(f"📝 Transcription: {data['transcription']}")
+                  print(f"🌍 Translation: {data['translation']}\n")
+
+          except websockets.exceptions.ConnectionClosed:
+              print("WebSocket connection closed.")
+
+  asyncio.run(listen())
+  ```
+
+## Development
+
+To contribute or modify this project, these steps might be helpful:
+
+**Clone** the repository:
+```bash
+git clone git@github.com:AbdullahHendy/live-translation.git
+cd live-translation
+```
+
+**Ceate** a virtual environment:
+```bash
+python -m venv .venv
+source .venv/bin/activate 
+```
+
+**Install** Dependencies:
+```bash
+pip install --upgrade pip
+pip install -r requirements.txt
+```
+
+**Test** the package:
+```bash
+make test
+```
+
+**Build** the package:
+```bash
+make build
+```
+> **NOTE**: Building generates a ***.whl*** file that can be ***pip*** insalled in a new environment for testing
+
+**If needed**, run the program within the virtual environment:
+```bash
+python -m live_translation.cli [OPTIONS]
+```
+
+## Tested Environment
+
+This project was tested and developed on the following system configuration:
+
+- **Architecture**: x86_64 (64-bit)
+- **Operating System**: Ubuntu 24.10 (Oracular Oriole)
+- **Kernel Version**: 6.11.0-18-generic
+- **Python Version**: 3.12.7
+- **Processor**: 13th Gen Intel(R) Core(TM) i9-13900HX
+- **GPU**: GeForce RTX 4070 Max-Q / Mobile [^1]
+- **RAM**: 16GB DDR5
+- **Dependencies**: All required dependencies are listed in `requirements.txt` and [Prerequisites](#prerequisites)
+
+[^1]: CUDA not utilized, as the `DEVICE` configuration is set to `"cpu"`. Additional Nvidia drivers, CUDA, cuDNN installation needed if option `"cuda"` were to be used.
+
+## Improvements
+
+- **Block Diagram**: Include a block diagram to visually represent the flow and architecture of the system, making it easier to understand the overall design.
+- **Better Error Handling**: Improve error handling across various components (audio, transcription, translation) to ensure the system is robust and can handle unexpected scenarios gracefully.
+- **Performance Optimization**: Investigate performance bottlenecks including checking sleep durations and optimizing concurrency management to minimize lag.
+- **Concurrency Design Check**: Review and optimize the threading design to ensure thread safety and prevent issues like race conditions or deadlocks, etc., revisit the current design of ***AudioRecorder*** being a thread while ***AudioProcessor***, ***Transcriber***, and ***Translator*** being processes.
+- **Logging**: Integrate detailed logging to track system activity, errors, and performance metrics using a more formal logging framework.
+
+## Citations
+ ```bibtex
+  @article{Whisper,
+    title = {Robust Speech Recognition via Large-Scale Weak Supervision},
+    url = {https://arxiv.org/abs/2212.04356},
+    author = {Radford, Alec and Kim, Jong Wook and Xu, Tao and Brockman, Greg and McLeavey, Christine and Sutskever, Ilya},
+    publisher = {arXiv},
+    year = {2022}
+  }
+
+  @misc{Silero VAD,
+    author = {Silero Team},
+    title = {Silero VAD: pre-trained enterprise-grade Voice Activity Detector (VAD), Number Detector and Language Classifier},
+    year = {2021},
+    publisher = {GitHub},
+    journal = {GitHub repository},
+    howpublished = {\url{https://github.com/snakers4/silero-vad}},
+    email = {hello@silero.ai}
+  }
+
+  @article{tiedemann2023democratizing,
+    title={Democratizing neural machine translation with {OPUS-MT}},
+    author={Tiedemann, J{\"o}rg and Aulamo, Mikko and Bakshandaeva, Daria and Boggia, Michele and Gr{\"o}nroos, Stig-Arne and Nieminen, Tommi and Raganato, Alessandro and Scherrer, Yves and Vazquez, Raul and Virpioja, Sami},
+    journal={Language Resources and Evaluation},
+    number={58},
+    pages={713--755},
+    year={2023},
+    publisher={Springer Nature},
+    issn={1574-0218},
+    doi={10.1007/s10579-023-09704-w}
+  }
+
+  @InProceedings{TiedemannThottingal:EAMT2020,
+    author = {J{\"o}rg Tiedemann and Santhosh Thottingal},
+    title = {{OPUS-MT} — {B}uilding open translation services for the {W}orld},
+    booktitle = {Proceedings of the 22nd Annual Conference of the European Association for Machine Translation (EAMT)},
+    year = {2020},
+    address = {Lisbon, Portugal}
+  }
+```

live_translation-0.1.0/live_translation/__init__.py

@@ -0,0 +1,6 @@
+# __init__.py
+
+from .app import LiveTranslationApp
+from .config import Config
+
+__all__ = ["LiveTranslationApp", "Config"]