simulstream 0.1.0__py3-none-any.whl

simulstream-0.1.0.dist-info/METADATA

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+Metadata-Version: 2.4
+Name: simulstream
+Version: 0.1.0
+Summary: A server to run simultaneous/streaming experiments and demo
+Author-email: Marco Gaido <mgaido@fbk.eu>, FBK HLT-MT <mt@fbk.eu>
+License:                                  Apache License
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+Project-URL: GitHub, https://github.com/hlt-mt/simulstream
+Project-URL: Issues, https://github.com/hlt-mt/simulstream/issues
+Keywords: simultaneous,demo,websocket,streaming
+Requires-Python: >=3.9
+Description-Content-Type: text/markdown
+License-File: LICENSE
+Requires-Dist: pyyaml>6.0
+Requires-Dist: websockets
+Requires-Dist: torch
+Requires-Dist: librosa
+Provides-Extra: dev
+Requires-Dist: pytest==7.4.0; extra == "dev"
+Requires-Dist: flake8; extra == "dev"
+Requires-Dist: sphinx; extra == "dev"
+Requires-Dist: sphinx-rtd-theme; extra == "dev"
+Requires-Dist: build; extra == "dev"
+Requires-Dist: twine; extra == "dev"
+Provides-Extra: hf
+Requires-Dist: transformers==4.48.1; extra == "hf"
+Provides-Extra: canary
+Requires-Dist: Cython; extra == "canary"
+Requires-Dist: nemo_toolkit[asr]==2.4.0; extra == "canary"
+Provides-Extra: vad
+Requires-Dist: silero-vad; extra == "vad"
+Provides-Extra: eval
+Requires-Dist: unbabel-comet==2.2.6; extra == "eval"
+Requires-Dist: mweralign; extra == "eval"
+Requires-Dist: sacrebleu; extra == "eval"
+Dynamic: license-file
+
+<div align="center">
+  <img src="https://github.com/hlt-mt/simulstream/blob/main/docs/source/_static/logo.png?raw=true" width="200" alt="simulstream logo">
+</div>
+
+
+# simulstream
+
+``simulstream`` is a Python library for simultaneous/streaming speech recognition and translation.
+It enables both the simulation with existing files to score systems, like in the
+[SimulEval](https://github.com/facebookresearch/SimulEval) project, and the possibility to run
+demos on a browser.
+
+``simulstream`` provides a WebSocket server and utilities for running streaming speech processing
+experiments and demos. It supports real-time transcription and translation through streaming audio
+input. By streaming, we mean that the library by default assumes that the input is an unbounded
+speech signal, rather than many short speech segments as in simultaneous speech processing.
+The simultaneous setting can be easily addressed by pre-segmenting the audio into many small
+segments and feed each segment to ``simulstream``.
+
+The repository is tested using Python 3.11. Although it may work also with other Python versions,
+we do not ensure compatibility with them. Check out the [Usage](#Usage) section for instructions on
+how to use the repository and the [Installation](#Installation) section for further information
+about how to install the project.
+
+
+## Installation
+
+You can install the latest stable version from PyPI:
+
+```shell
+pip install simulstream
+```
+
+Or, to install from source:
+
+```shell
+git clone https://github.com/hlt-mt/simulstream.git
+cd simulstream
+pip install .
+```
+
+Please notice that these commands will only install the basic functionalities of the repository,
+i.e. the WebSocket server and client. Additionally, you have to install the dependencies required
+by the `speech processor` that you want to use. The repository comes with examples of speech
+processors that rely on e.g. Transformers models or on the NVIDIA Canary model. You can install
+the required dependencies for them by specifying the corresponding selector when installing
+the repository. For instance, using Canary speech processors requires the ``canary`` selector.
+But you can create your custom speech processor and install the corresponding dependencies.
+
+Also the evaluation comes with additional dependencies, which can be installed with the ``eval``
+selector. Be careful as the metrics include COMET, which has dependencies on Transformers and
+other libraries that can run on conflict with those required by your speech processor.
+
+As an example, if you want to install the ``simulstream`` package with Canary speech processors
+and the evaluation package, run:
+
+```shell
+pip install -e .[canary,eval]
+```
+
+For development (with docs and testing tools):
+
+```shell
+pip install -e .[dev]
+```
+
+## Usage
+
+The package is based on a WebSocket client-server interaction. The server waits for connections
+by the clients. The clients can send configurations messages in JSON format (e.g. to specify
+the input/output desired language) and audio chunks, encoded as 16-bit integers. The server
+processes the input audio, generates the transcript/translation by means of the configured
+speech processor and sends it to the client in JSON format. In addition, if configured to do so,
+the server writes logs that can be used to compute metrics in a JSONL file.
+
+To have everything up and running, first run the server and once the server is up and running
+execute one or more clients. The repository contains the code for two types of clients: a web
+interface client that can be used for demos and a python command-line client that can be used
+for testing or evaluation of the speech processor performance.
+
+Below, you can find a simple illustration of the overall architecture with the two scenarios:
+
+<img src="docs/source/_static/architecture.png" alt="Overall simulstream architecture">
+
+To streamline the evaluation of speech processors in terms of quality and latency, there is also
+the possibility to run the inference with a configured speech processor without setting up a
+client-server interaction. For more information, refer to the [Evaluation section](#Evaluation).
+
+### WebSocket Server
+
+Run the WebSocket server with YAML configuration files:
+
+```shell
+simulstream_server --server-config config/server.yaml \
+    --speech-processor-config config/speech_processor.yaml
+```
+
+The repository contains examples of YAML files both for the server and for some speech processors.
+They can be edited and adapted.
+
+#### Customize with Your Speech Processor
+
+If you want to implement your own ``speech_processor``, you need to create a subclass
+of ``simulstream.server.speech_processors.base.SpeechProcessor`` and add the class
+to the ``PYTHONPATH``. Then reference the new class in the YAML file configuration
+of the speech processor. Refer to the dedicated documentation for more
+details on how to implement a speech processor in the project documentation and in the
+docstring of both [``simulstream.server.speech_processors.base.SpeechProcessor``](simulstream/server/speech_processors/__init__.py)
+and [`simulstream.server.speech_processors.base.BaseSpeechProcessor`](simulstream/server/speech_processors/base.py).
+
+As reference, you can check the processors implemented in this repository, available in the
+[`simulstream.server.speech_processors`](simulstream/server/speech_processors) module.
+Notice that each speech processor can have additional dependencies that  are not installed by
+default with this codebase (see [Installation](#installation)).
+
+
+### HTTP Server Web Client
+
+For a demo, you can create an HTTP web server that servers a web interface interacting with the
+WebSocket server. This can be done by:
+
+```shell
+simulstream_demo_http_server --config config/server.yaml -p 8001 --directory webdemo
+```
+
+In case you are not running the command from the root folder of this repository, you should change
+accordingly the ``--directory`` parameter to point to the ``webdemo`` folder containing the HTML
+files of the HTTP demo. You can of course replace ``8001`` with any other port number you prefer.
+The web interface can then be accessed from the local laptop at ``http://localhost:8001/`` or from
+any other terminal connected  to the same network using the IP address of your workstation instead
+of ``localhost``. Be careful not to use the same port specified for the WebSocket server if they
+are running on the same machine. If running the HTTP server from a machine different from the one
+where the WebSocket server runs, ensure that the HTTP server can connect to the WebSocket server
+through the address specified in the ``config/server.yaml`` file.
+
+
+### Python Client
+
+
+If you want to process a set of audio files (e.g. a test set) with your system, send the audios
+with the provided client:
+
+```shell
+simulstream_wavs_client --uri ws://localhost:8080/ \
+    --wav-list-file PATH_TO_TXT_FILE_WITH_A_LIST_OF_WAV_FILES.txt \
+    --tgt-lang it --src-lang en
+```
+
+Tne ``--uri`` should contain the address of the WebSocket client, so it should correspond to what
+is specified in the server YAML config. The file specified in ``--wav-list-file`` should be a TXT
+files containing for each line the path to a WAV file.
+
+Before running the command, ensure that the logging of metrics is enabled in the configuration file
+of the WebSocket server (i.e., ``metrics.enabled = True``) and that metrics are logged to an
+empty/non-existing file, to ensure that the resulting file will contain only the logs related to
+the files sent by the client. Then, compute the relevant scores as described below
+([Evaluation](#Evaluation)).
+
+
+### Evaluation
+
+To evaluate a speech processor, as a first thing you need to run the inference on a test set.
+This can be done either through the Python client (see section above) or through the dedicated
+command that performs the inference without setting up a client-server interaction. This second
+option can be followed by running the dedicated command:
+
+```
+simulstream_inference --speech-processor-config config/speech_processor.yaml \
+    --wav-list-file PATH_TO_TXT_FILE_WITH_A_LIST_OF_WAV_FILES.txt \
+    --tgt-lang it --src-lang en --metrics-log-file metrics.jsonl
+```
+
+Once you have generated the JSONL file containing the inference log (e.g. ``metrics.jsonl``), you
+can score your speech processor by running:
+
+```shell
+simulstream_score_latency --scorer stream_laal \
+    --eval-config config/speech_processor.yaml \
+    --log-file metrics.jsonl \
+    --reference REFERENCE_FILE.txt \
+    --audio-definition YAML_AUDIO_REFERENCES_DEFINITION.yaml
+
+simulstream_score_quality --scorer comet \
+    --eval-config config/speech_processor.yaml \
+    --log-file metrics.jsonl \
+    --references REFERENCES_FILE.txt \
+    --transcripts TRANSCRIPTS_FILE.txt
+
+simulstream_stats --eval-config config/speech_processor.yaml \
+    --log-file metrics.jsonl
+```
+
+Each of them will output different metrics. ``simulstream_score_latency`` provides the metric for
+the latency of the system by leveraging a sentence-based TXT reference file and a YAML file, which
+for each line in the reference TXT contains the WAV file to which it belong, its ``offset``
+(i.e. start time) and ``duration``. The exact definition of how the latency is defined depends on
+the selected metric (``--scorer``).
+
+Similarly, ``simulstream_score_quality`` evaluated the quality
+of the generated outputs against one (or more) reference (and transcript, only for metrics
+requiring them) file(s).
+
+Lastly, ``simulstream_stats`` computes statistics like the computational cost and flickering ratio.
+
+
+## Contributing
+
+Contributions from interested researchers and developers are extremely appreciated.
+
+You can create an ***issue*** in case of problems with the code, questions, or feature requests.
+You are also more than welcome to create a ***pull request*** that addresses any ***issue***.
+
+## Licence
+
+``simulstream`` is licensed under [Apache Version 2.0](LICENSE). 
+
+## Credits
+If you use this library, please cite:
+
+
+```
+@article{gaido-et-al-2025-simulstream,
+    title={{simulstream: Open-Source Toolkit for Evaluation and Demonstration of Streaming Speech-to-Text Translation Systems}},
+    author={Gaido, Marco and Papi, Sara and Cettolo, Mauro and Negri, Matteo and Bentivogli, Luisa},
+    journal = "arXiv",
+    year={2025}
+}
+```

simulstream-0.1.0.dist-info/RECORD

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+simulstream/server/speech_processors/incremental_output.py,sha256=DVIaHCGTF4v2PzqUHj30Xr-4RtJG6BOqpAMLJY5yG4U,3383
+simulstream/server/speech_processors/seamless_sliding_window_retranslation.py,sha256=G52qTfIm4qiDYUNf1tbdOcJCKE3u-I9KCjpbVSZLorg,3642
+simulstream/server/speech_processors/seamless_streamatt.py,sha256=_Z01FKPJEOSPX2yLHwoBYVebAeagyhSCMbQwnex0h-A,12618
+simulstream/server/speech_processors/simuleval_wrapper.py,sha256=K-gljTmlkJJ9C_RDo34DzNV-dxJdo1xpP3s9mGFzZs0,6655
+simulstream/server/speech_processors/sliding_window_retranslation.py,sha256=h-zbF1Ydy41y34aAZKrmO0NVCh_jGhXYQ1Qdm0Bvxqc,6653
+simulstream/server/speech_processors/vad_wrapper.py,sha256=Ecd0dr_xHxl9kKdswR1-OcEZdPgH9HmEDzz0S6Okrzw,9251
+simulstream-0.1.0.dist-info/licenses/LICENSE,sha256=knu-Hk-hl9xNWU_Gn_6Msa5U8gHhDT8yIHW7_KrmdhE,11333
+uts/__init__.py,sha256=47DEQpj8HBSa-_TImW-5JCeuQeRkm5NMpJWZG3hSuFU,0
+uts/utils.py,sha256=CXC0tahql0HZN4Cb2sQLyYsaW22sDz3n8gCvcVFyx2c,127
+uts/metrics/__init__.py,sha256=47DEQpj8HBSa-_TImW-5JCeuQeRkm5NMpJWZG3hSuFU,0
+uts/metrics/log_reader.py,sha256=6xVFeGDJjl_66juMWl9M5itTZjGv6swowhnpaUwUFcU,2195
+uts/speech_processors/__init__.py,sha256=47DEQpj8HBSa-_TImW-5JCeuQeRkm5NMpJWZG3hSuFU,0
+uts/speech_processors/test_simuleval_wrapper.py,sha256=50U9a0DVc-X4ySYsrAooy4wYNM-Uwg0A2PnnMUQNTY8,3517
+simulstream-0.1.0.dist-info/METADATA,sha256=gaFMBeG_4LSuccVtuIaxHKGmxCKyhq_vx5TnduuLXI0,24508
+simulstream-0.1.0.dist-info/WHEEL,sha256=_zCd3N1l69ArxyTb8rzEoP9TpbYXkqRFSNOD5OuxnTs,91
+simulstream-0.1.0.dist-info/entry_points.txt,sha256=5j84rLKVesw47qjaeELpz25XfzuSg_Zz0JLhcO3Y6Gg,479
+simulstream-0.1.0.dist-info/top_level.txt,sha256=DF787K6iu65z7cBQrCRze-l8w8s-ouExTI85-Vxyka8,21
+simulstream-0.1.0.dist-info/RECORD,,

simulstream-0.1.0.dist-info/WHEEL

@@ -0,0 +1,5 @@
+Wheel-Version: 1.0
+Generator: setuptools (80.9.0)
+Root-Is-Purelib: true
+Tag: py3-none-any
+

simulstream-0.1.0.dist-info/entry_points.txt

@@ -0,0 +1,8 @@
+[console_scripts]
+simulstream_demo_http_server = simulstream.server.http_server:cli_main
+simulstream_inference = simulstream.inference:cli_main
+simulstream_score_latency = simulstream.metrics.score_latency:cli_main
+simulstream_score_quality = simulstream.metrics.score_quality:cli_main
+simulstream_server = simulstream.server.websocket_server:cli_main
+simulstream_stats = simulstream.metrics.stats:cli_main
+simulstream_wavs_client = simulstream.client.wav_reader_client:cli_main