add new project

CONTRIBUTING.md

+# Contributing to faster-whisper
+
+Contributions are welcome! Here are some pointers to help you install the library for development and validate your changes before submitting a pull request.
+
+## Install the library for development
+
+We recommend installing the module in editable mode with the `dev` extra requirements:
+
+```bash
+git clone https://github.com/SYSTRAN/faster-whisper.git
+cd faster-whisper/
+pip install -e .[dev]
+```
+
+## Validate the changes before creating a pull request
+
+1. Make sure the existing tests are still passing (and consider adding new tests as well!):
+
+```bash
+pytest tests/
+```
+
+2. Reformat and validate the code with the following tools:
+
+```bash
+black .
+isort .
+flake8 .
+```
+
+These steps are also run automatically in the CI when you open the pull request.

LICENSE

+MIT License
+
+Copyright (c) 2023 SYSTRAN
+
+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.

MANIFEST.in

+include faster_whisper/assets/silero_vad.onnx
+include requirements.txt
+include requirements.conversion.txt
+include faster_whisper/assets/pyannote_vad_model.bin

README.md

+[![CI](https://github.com/SYSTRAN/faster-whisper/workflows/CI/badge.svg)](https://github.com/SYSTRAN/faster-whisper/actions?query=workflow%3ACI) [![PyPI version](https://badge.fury.io/py/faster-whisper.svg)](https://badge.fury.io/py/faster-whisper)
+
+# Faster Whisper transcription with CTranslate2
+
+**faster-whisper** is a reimplementation of OpenAI's Whisper model using [CTranslate2](https://github.com/OpenNMT/CTranslate2/), which is a fast inference engine for Transformer models.
+
+This implementation is up to 4 times faster than [openai/whisper](https://github.com/openai/whisper) for the same accuracy while using less memory. The efficiency can be further improved with 8-bit quantization on both CPU and GPU.
+
+## Benchmark
+
+### Whisper
+
+For reference, here's the time and memory usage that are required to transcribe [**13 minutes**](https://www.youtube.com/watch?v=0u7tTptBo9I) of audio using different implementations:
+
+* [openai/whisper](https://github.com/openai/whisper)@[6dea21fd](https://github.com/openai/whisper/commit/6dea21fd7f7253bfe450f1e2512a0fe47ee2d258)
+* [whisper.cpp](https://github.com/ggerganov/whisper.cpp)@[3b010f9](https://github.com/ggerganov/whisper.cpp/commit/3b010f9bed9a6068609e9faf52383aea792b0362)
+* [faster-whisper](https://github.com/SYSTRAN/faster-whisper)@[cce6b53e](https://github.com/SYSTRAN/faster-whisper/commit/cce6b53e4554f71172dad188c45f10fb100f6e3e)
+
+### Large-v2 model on GPU
+
+| Implementation | Precision | Beam size | Time | Max. GPU memory | Max. CPU memory |
+| --- | --- | --- | --- | --- | --- |
+| openai/whisper | fp16 | 5 | 4m30s | 11325MB | 9439MB |
+| faster-whisper | fp16 | 5 | 54s | 4755MB | 3244MB |
+| faster-whisper | int8 | 5 | 59s | 3091MB | 3117MB |
+
+*Executed with CUDA 11.7.1 on a NVIDIA Tesla V100S.*
+
+### Small model on CPU
+
+| Implementation | Precision | Beam size | Time | Max. memory |
+| --- | --- | --- | --- | --- |
+| openai/whisper | fp32 | 5 | 10m31s | 3101MB |
+| whisper.cpp | fp32 | 5 | 17m42s | 1581MB |
+| whisper.cpp | fp16 | 5 | 12m39s | 873MB |
+| faster-whisper | fp32 | 5 | 2m44s | 1675MB |
+| faster-whisper | int8 | 5 | 2m04s | 995MB |
+
+*Executed with 8 threads on a Intel(R) Xeon(R) Gold 6226R.*
+
+
+### Distil-whisper
+
+| Implementation | Precision | Beam size | Time | Gigaspeech WER |
+| --- | --- | --- | --- | --- |
+| distil-whisper/distil-large-v2 | fp16 | 4 |- | 10.36 |
+| [faster-distil-large-v2](https://huggingface.co/Systran/faster-distil-whisper-large-v2) | fp16 | 5 | - | 10.28 |
+| distil-whisper/distil-medium.en | fp16 | 4 | - | 11.21 |
+| [faster-distil-medium.en](https://huggingface.co/Systran/faster-distil-whisper-medium.en) | fp16 | 5 | - | 11.21 |
+
+*Executed with CUDA 11.4 on a NVIDIA 3090.*
+
+<details>
+<summary>testing details (click to expand)</summary>
+
+For `distil-whisper/distil-large-v2`, the WER is tested with code sample from [link](https://huggingface.co/distil-whisper/distil-large-v2#evaluation). for `faster-distil-whisper`, the WER is tested with setting:
+```python
+from faster_whisper import WhisperModel
+
+model_size = "distil-large-v2"
+# model_size = "distil-medium.en"
+# Run on GPU with FP16
+model = WhisperModel(model_size, device="cuda", compute_type="float16")
+segments, info = model.transcribe("audio.mp3", beam_size=5, language="en")
+```
+</details>
+
+## Requirements
+
+* Python 3.8 or greater
+
+
+### GPU
+
+GPU execution requires the following NVIDIA libraries to be installed:
+
+* [cuBLAS for CUDA 12](https://developer.nvidia.com/cublas)
+* [cuDNN 8 for CUDA 12](https://developer.nvidia.com/cudnn)
+
+**Note**: Latest versions of `ctranslate2` support CUDA 12 only. For CUDA 11, the current workaround is downgrading to the `3.24.0` version of `ctranslate2` (This can be done with `pip install --force-reinstall ctranslate2==3.24.0` or specifying the version in a `requirements.txt`).
+
+There are multiple ways to install the NVIDIA libraries mentioned above. The recommended way is described in the official NVIDIA documentation, but we also suggest other installation methods below. 
+
+<details>
+<summary>Other installation methods (click to expand)</summary>
+
+
+**Note:** For all these methods below, keep in mind the above note regarding CUDA versions. Depending on your setup, you may need to install the _CUDA 11_ versions of libraries that correspond to the CUDA 12 libraries listed in the instructions below.
+
+#### Use Docker
+
+The libraries (cuBLAS, cuDNN) are installed in these official NVIDIA CUDA Docker images: `nvidia/cuda:12.0.0-runtime-ubuntu20.04` or `nvidia/cuda:12.0.0-runtime-ubuntu22.04`.
+
+#### Install with `pip` (Linux only)
+
+On Linux these libraries can be installed with `pip`. Note that `LD_LIBRARY_PATH` must be set before launching Python.
+
+```bash
+pip install nvidia-cublas-cu12 nvidia-cudnn-cu12
+
+export LD_LIBRARY_PATH=`python3 -c 'import os; import nvidia.cublas.lib; import nvidia.cudnn.lib; print(os.path.dirname(nvidia.cublas.lib.__file__) + ":" + os.path.dirname(nvidia.cudnn.lib.__file__))'`
+```
+
+**Note**: Version 9+ of `nvidia-cudnn-cu12` appears to cause issues due its reliance on cuDNN 9 (Faster-Whisper does not currently support cuDNN 9). Ensure your version of the Python package is for cuDNN 8.
+
+#### Download the libraries from Purfview's repository (Windows & Linux)
+
+Purfview's [whisper-standalone-win](https://github.com/Purfview/whisper-standalone-win) provides the required NVIDIA libraries for Windows & Linux in a [single archive](https://github.com/Purfview/whisper-standalone-win/releases/tag/libs). Decompress the archive and place the libraries in a directory included in the `PATH`.
+
+</details>
+
+## Installation
+
+The module can be installed from [PyPI](https://pypi.org/project/faster-whisper/):
+
+```bash
+pip install faster-whisper
+```
+
+<details>
+<summary>Other installation methods (click to expand)</summary>
+
+### Install the master branch
+
+```bash
+pip install --force-reinstall "faster-whisper @ https://github.com/SYSTRAN/faster-whisper/archive/refs/heads/master.tar.gz"
+```
+
+### Install a specific commit
+
+```bash
+pip install --force-reinstall "faster-whisper @ https://github.com/SYSTRAN/faster-whisper/archive/a4f1cc8f11433e454c3934442b5e1a4ed5e865c3.tar.gz"
+```
+
+</details>
+
+## Usage
+
+### Faster-whisper
+
+```python
+from faster_whisper import WhisperModel
+
+model_size = "large-v3"
+
+# Run on GPU with FP16
+model = WhisperModel(model_size, device="cuda", compute_type="float16")
+
+# or run on GPU with INT8
+# model = WhisperModel(model_size, device="cuda", compute_type="int8_float16")
+# or run on CPU with INT8
+# model = WhisperModel(model_size, device="cpu", compute_type="int8")
+
+segments, info = model.transcribe("audio.mp3", beam_size=5)
+
+print("Detected language '%s' with probability %f" % (info.language, info.language_probability))
+
+for segment in segments:
+    print("[%.2fs -> %.2fs] %s" % (segment.start, segment.end, segment.text))
+```
+
+**Warning:** `segments` is a *generator* so the transcription only starts when you iterate over it. The transcription can be run to completion by gathering the segments in a list or a `for` loop:
+
+```python
+segments, _ = model.transcribe("audio.mp3")
+segments = list(segments)  # The transcription will actually run here.
+```
+
+### multi-segment language detection
+
+To directly use the model for improved language detection, the following code snippet can be used:
+
+```python
+from faster_whisper import WhisperModel
+model = WhisperModel("medium", device="cuda", compute_type="float16")
+language_info = model.detect_language_multi_segment("audio.mp3")
+```
+
+### Batched faster-whisper
+
+
+The batched version of faster-whisper is inspired by [whisper-x](https://github.com/m-bain/whisperX) licensed under the BSD-2 Clause license and integrates its VAD model to this library. We modify this implementation and also replaced the feature extraction with a faster torch-based implementation. Batched version improves the speed upto 10-12x compared to openAI implementation and 3-4x compared to the sequential faster_whisper version. It works by transcribing semantically meaningful audio chunks as batches leading to faster inference. 
+
+The following code snippet illustrates how to run inference with batched version on an example audio file. Please also refer to the test scripts of batched faster whisper.
+
+```python
+from faster_whisper import WhisperModel, BatchedInferencePipeline
+
+model = WhisperModel("medium", device="cuda", compute_type="float16")
+batched_model = BatchedInferencePipeline(model=model)
+segments, info = batched_model.transcribe("audio.mp3", batch_size=16)
+
+for segment in segments:
+    print("[%.2fs -> %.2fs] %s" % (segment.start, segment.end, segment.text))
+```
+
+### Faster Distil-Whisper
+
+The Distil-Whisper checkpoints are compatible with the Faster-Whisper package. In particular, the latest [distil-large-v3](https://huggingface.co/distil-whisper/distil-large-v3)
+checkpoint is intrinsically designed to work with the Faster-Whisper transcription algorithm. The following code snippet 
+demonstrates how to run inference with distil-large-v3 on a specified audio file:
+
+```python
+from faster_whisper import WhisperModel
+
+model_size = "distil-large-v3"
+
+model = WhisperModel(model_size, device="cuda", compute_type="float16")
+segments, info = model.transcribe("audio.mp3", beam_size=5, language="en", condition_on_previous_text=False)
+
+for segment in segments:
+    print("[%.2fs -> %.2fs] %s" % (segment.start, segment.end, segment.text))
+```
+
+For more information about the distil-large-v3 model, refer to the original [model card](https://huggingface.co/distil-whisper/distil-large-v3).
+
+### Word-level timestamps
+
+```python
+segments, _ = model.transcribe("audio.mp3", word_timestamps=True)
+
+for segment in segments:
+    for word in segment.words:
+        print("[%.2fs -> %.2fs] %s" % (word.start, word.end, word.word))
+```
+
+### VAD filter
+
+The library integrates the [Silero VAD](https://github.com/snakers4/silero-vad) model to filter out parts of the audio without speech:
+
+```python
+segments, _ = model.transcribe("audio.mp3", vad_filter=True)
+```
+
+The default behavior is conservative and only removes silence longer than 2 seconds. See the available VAD parameters and default values in the [source code](https://github.com/SYSTRAN/faster-whisper/blob/master/faster_whisper/vad.py). They can be customized with the dictionary argument `vad_parameters`:
+
+```python
+segments, _ = model.transcribe(
+    "audio.mp3",
+    vad_filter=True,
+    vad_parameters=dict(min_silence_duration_ms=500),
+)
+```
+
+### Logging
+
+The library logging level can be configured like this:
+
+```python
+import logging
+
+logging.basicConfig()
+logging.getLogger("faster_whisper").setLevel(logging.DEBUG)
+```
+
+### Going further
+
+See more model and transcription options in the [`WhisperModel`](https://github.com/SYSTRAN/faster-whisper/blob/master/faster_whisper/transcribe.py) class implementation.
+
+## Community integrations
+
+Here is a non exhaustive list of open-source projects using faster-whisper. Feel free to add your project to the list!
+
+
+* [faster-whisper-server](https://github.com/fedirz/faster-whisper-server) is an OpenAI compatible server using `faster-whisper`. It's easily deployable with Docker, works with OpenAI SDKs/CLI, supports streaming, and live transcription.
+* [WhisperX](https://github.com/m-bain/whisperX) is an award-winning Python library that offers speaker diarization and accurate word-level timestamps using wav2vec2 alignment
+* [whisper-ctranslate2](https://github.com/Softcatala/whisper-ctranslate2) is a command line client based on faster-whisper and compatible with the original client from openai/whisper.
+* [whisper-diarize](https://github.com/MahmoudAshraf97/whisper-diarization) is a speaker diarization tool that is based on faster-whisper and NVIDIA NeMo.
+* [whisper-standalone-win](https://github.com/Purfview/whisper-standalone-win) Standalone CLI executables of faster-whisper for Windows, Linux & macOS. 
+* [asr-sd-pipeline](https://github.com/hedrergudene/asr-sd-pipeline) provides a scalable, modular, end to end multi-speaker speech to text solution implemented using AzureML pipelines.
+* [Open-Lyrics](https://github.com/zh-plus/Open-Lyrics) is a Python library that transcribes voice files using faster-whisper, and translates/polishes the resulting text into `.lrc` files in the desired language using OpenAI-GPT.
+* [wscribe](https://github.com/geekodour/wscribe) is a flexible transcript generation tool supporting faster-whisper, it can export word level transcript and the exported transcript then can be edited with [wscribe-editor](https://github.com/geekodour/wscribe-editor)
+* [aTrain](https://github.com/BANDAS-Center/aTrain) is a graphical user interface implementation of faster-whisper developed at the BANDAS-Center at the University of Graz for transcription and diarization in Windows ([Windows Store App](https://apps.microsoft.com/detail/atrain/9N15Q44SZNS2)) and Linux.
+* [Whisper-Streaming](https://github.com/ufal/whisper_streaming) implements real-time mode for offline Whisper-like speech-to-text models with faster-whisper as the most recommended back-end. It implements a streaming policy with self-adaptive latency based on the actual source complexity, and demonstrates the state of the art.
+* [WhisperLive](https://github.com/collabora/WhisperLive) is a nearly-live implementation of OpenAI's Whisper which uses faster-whisper as the backend to transcribe audio in real-time.
+* [Faster-Whisper-Transcriber](https://github.com/BBC-Esq/ctranslate2-faster-whisper-transcriber) is a simple but reliable voice transcriber that provides a user-friendly interface.
+
+## Model conversion
+
+When loading a model from its size such as `WhisperModel("large-v3")`, the corresponding CTranslate2 model is automatically downloaded from the [Hugging Face Hub](https://huggingface.co/Systran).
+
+We also provide a script to convert any Whisper models compatible with the Transformers library. They could be the original OpenAI models or user fine-tuned models.
+
+For example the command below converts the [original "large-v3" Whisper model](https://huggingface.co/openai/whisper-large-v3) and saves the weights in FP16:
+
+```bash
+pip install transformers[torch]>=4.23
+
+ct2-transformers-converter --model openai/whisper-large-v3 --output_dir whisper-large-v3-ct2
+--copy_files tokenizer.json preprocessor_config.json --quantization float16
+```
+
+* The option `--model` accepts a model name on the Hub or a path to a model directory.
+* If the option `--copy_files tokenizer.json` is not used, the tokenizer configuration is automatically downloaded when the model is loaded later.
+
+Models can also be converted from the code. See the [conversion API](https://opennmt.net/CTranslate2/python/ctranslate2.converters.TransformersConverter.html).
+
+### Load a converted model
+
+1. Directly load the model from a local directory:
+```python
+model = faster_whisper.WhisperModel("whisper-large-v3-ct2")
+```
+
+2. [Upload your model to the Hugging Face Hub](https://huggingface.co/docs/transformers/model_sharing#upload-with-the-web-interface) and load it from its name:
+```python
+model = faster_whisper.WhisperModel("username/whisper-large-v3-ct2")
+```
+
+## Comparing performance against other implementations
+
+If you are comparing the performance against other Whisper implementations, you should make sure to run the comparison with similar settings. In particular:
+
+* Verify that the same transcription options are used, especially the same beam size. For example in openai/whisper, `model.transcribe` uses a default beam size of 1 but here we use a default beam size of 5.
+* When running on CPU, make sure to set the same number of threads. Many frameworks will read the environment variable `OMP_NUM_THREADS`, which can be set when running your script:
+
+```bash
+OMP_NUM_THREADS=4 python3 my_script.py
+```

benchmark/memory_benchmark.py

+import argparse
+import time
+
+from typing import Callable
+
+import py3nvml.py3nvml as nvml
+
+from memory_profiler import memory_usage
+from utils import MyThread, get_logger, inference
+
+logger = get_logger("faster-whisper")
+parser = argparse.ArgumentParser(description="Memory benchmark")
+parser.add_argument(
+    "--gpu_memory", action="store_true", help="Measure GPU memory usage"
+)
+parser.add_argument("--device-index", type=int, default=0, help="GPU device index")
+parser.add_argument(
+    "--interval",
+    type=float,
+    default=0.5,
+    help="Interval at which measurements are collected",
+)
+args = parser.parse_args()
+device_idx = args.device_index
+interval = args.interval
+
+
+def measure_memory(func: Callable[[], None]):
+    if args.gpu_memory:
+        logger.info(
+            "Measuring maximum GPU memory usage on GPU device."
+            " Make sure to not have additional processes running on the same GPU."
+        )
+        # init nvml
+        nvml.nvmlInit()
+        handle = nvml.nvmlDeviceGetHandleByIndex(device_idx)
+        gpu_name = nvml.nvmlDeviceGetName(handle)
+        gpu_memory_limit = nvml.nvmlDeviceGetMemoryInfo(handle).total >> 20
+        gpu_power_limit = nvml.nvmlDeviceGetPowerManagementLimit(handle) / 1000.0
+        info = {"gpu_memory_usage": [], "gpu_power_usage": []}
+
+        def _get_gpu_info():
+            while True:
+                info["gpu_memory_usage"].append(
+                    nvml.nvmlDeviceGetMemoryInfo(handle).used >> 20
+                )
+                info["gpu_power_usage"].append(
+                    nvml.nvmlDeviceGetPowerUsage(handle) / 1000
+                )
+                time.sleep(interval)
+
+                if stop:
+                    break
+
+            return info
+
+        stop = False
+        thread = MyThread(_get_gpu_info, params=())
+        thread.start()
+        func()
+        stop = True
+        thread.join()
+        result = thread.get_result()
+
+        # shutdown nvml
+        nvml.nvmlShutdown()
+        max_memory_usage = max(result["gpu_memory_usage"])
+        max_power_usage = max(result["gpu_power_usage"])
+        print("GPU name: %s" % gpu_name)
+        print("GPU device index: %s" % device_idx)
+        print(
+            "Maximum GPU memory usage: %dMiB / %dMiB (%.2f%%)"
+            % (
+                max_memory_usage,
+                gpu_memory_limit,
+                (max_memory_usage / gpu_memory_limit) * 100,
+            )
+        )
+        print(
+            "Maximum GPU power usage: %dW / %dW (%.2f%%)"
+            % (
+                max_power_usage,
+                gpu_power_limit,
+                (max_power_usage / gpu_power_limit) * 100,
+            )
+        )
+    else:
+        logger.info("Measuring maximum increase of memory usage.")
+        max_usage = memory_usage(func, max_usage=True, interval=interval)
+        print("Maximum increase of RAM memory usage: %d MiB" % max_usage)
+
+
+if __name__ == "__main__":
+    measure_memory(inference)

benchmark/requirements.benchmark.txt

+transformers
+jiwer
+evaluate
+datasets
+memory_profiler
+py3nvml

benchmark/speed_benchmark.py

+import argparse
+import timeit
+
+from typing import Callable
+
+from utils import inference
+
+parser = argparse.ArgumentParser(description="Speed benchmark")
+parser.add_argument(
+    "--repeat",
+    type=int,
+    default=3,
+    help="Times an experiment will be run.",
+)
+args = parser.parse_args()
+
+
+def measure_speed(func: Callable[[], None]):
+    # as written in https://docs.python.org/3/library/timeit.html#timeit.Timer.repeat,
+    # min should be taken rather than the average
+    runtimes = timeit.repeat(
+        func,
+        repeat=args.repeat,
+        number=10,
+    )
+    print(runtimes)
+    print("Min execution time: %.3fs" % (min(runtimes) / 10.0))
+
+
+if __name__ == "__main__":
+    measure_speed(inference)

benchmark/utils.py

+import logging
+
+from threading import Thread
+from typing import Optional
+
+from faster_whisper import WhisperModel
+
+model_path = "large-v3"
+model = WhisperModel(model_path, device="cuda")
+
+
+def inference():
+    segments, info = model.transcribe("benchmark.m4a", language="fr")
+    for segment in segments:
+        print("[%.2fs -> %.2fs] %s" % (segment.start, segment.end, segment.text))
+
+
+def get_logger(name: Optional[str] = None) -> logging.Logger:
+    formatter = logging.Formatter("%(levelname)s: %(message)s")
+    logger = logging.getLogger(name)
+    logger.setLevel(logging.DEBUG)
+    handler = logging.StreamHandler()
+    handler.setFormatter(formatter)
+    logger.addHandler(handler)
+    return logger
+
+
+class MyThread(Thread):
+    def __init__(self, func, params):
+        super(MyThread, self).__init__()
+        self.func = func
+        self.params = params
+        self.result = None
+
+    def run(self):
+        self.result = self.func(*self.params)
+
+    def get_result(self):
+        return self.result

benchmark/wer_benchmark.py

+import argparse
+import json
+import os
+
+from datasets import load_dataset
+from evaluate import load
+from tqdm import tqdm
+from transformers.models.whisper.english_normalizer import EnglishTextNormalizer
+
+from faster_whisper import WhisperModel
+
+parser = argparse.ArgumentParser(description="WER benchmark")
+parser.add_argument(
+    "--audio_numb",
+    type=int,
+    default=None,
+    help="Specify the number of validation audio files in the dataset."
+    " Set to None to retrieve all audio files.",
+)
+args = parser.parse_args()
+
+model_path = "large-v3"
+model = WhisperModel(model_path, device="cuda")
+
+# load the dataset with streaming mode
+dataset = load_dataset("librispeech_asr", "clean", split="validation", streaming=True)
+
+# define the evaluation metric
+wer_metric = load("wer")
+
+with open(os.path.join(os.path.dirname(__file__), "normalizer.json"), "r") as f:
+    normalizer = EnglishTextNormalizer(json.load(f))
+
+
+def inference(batch):
+    batch["transcription"] = []
+    for sample in batch["audio"]:
+        segments, info = model.transcribe(sample["array"], language="en")
+        batch["transcription"].append("".join([segment.text for segment in segments]))
+    batch["reference"] = batch["text"]
+    return batch
+
+
+dataset = dataset.map(function=inference, batched=True, batch_size=16)
+
+all_transcriptions = []
+all_references = []
+
+# iterate over the dataset and run inference
+for i, result in tqdm(enumerate(dataset), desc="Evaluating..."):
+    all_transcriptions.append(result["transcription"])
+    all_references.append(result["reference"])
+    if args.audio_numb and i == (args.audio_numb - 1):
+        break
+
+# normalize predictions and references
+all_transcriptions = [normalizer(transcription) for transcription in all_transcriptions]
+all_references = [normalizer(reference) for reference in all_references]
+
+# compute the WER metric
+wer = 100 * wer_metric.compute(
+    predictions=all_transcriptions, references=all_references
+)
+print("WER: %.3f" % wer)

demo.py

+from faster_whisper import WhisperModel
+import time
+model_size = "small"
+
+path = "/modelpath/models--Systran--faster-whisper-small/"
+
+# Run on GPU with FP16
+model = WhisperModel(model_size_or_path=path, device="cuda", local_files_only=True)
+
+# or run on GPU with INT8
+# model = WhisperModel(model_size, device="cuda", compute_type="int8_float16")
+# or run on CPU with INT8
+# model = WhisperModel(model_size, device="cpu", compute_type="int8")
+
+start_time = time.time()
+
+segments, info = model.transcribe("./test1.mp4", beam_size=5, language="zh", vad_filter=True, vad_parameters=dict(min_silence_duration_ms=1000))
+
+print("Detected language '%s' with probability %f" % (info.language, info.language_probability))
+
+for segment in segments:
+    print("[%.2fs -> %.2fs] %s" % (segment.start, segment.end, segment.text))
+
+end_time = time.time()
+print('time:',end_time - start_time)

docker/Dockerfile

+FROM nvidia/cuda:12.2.2-cudnn8-runtime-ubuntu22.04
+WORKDIR /root
+RUN apt-get update -y && apt-get install -y python3-pip
+COPY infer.py jfk.flac ./
+RUN pip3 install faster-whisper
+CMD ["python3", "infer.py"]

docker/infer.py

+from faster_whisper import WhisperModel
+
+jfk_path = "jfk.flac"
+model = WhisperModel("tiny", device="cuda")
+segments, info = model.transcribe(jfk_path, word_timestamps=True)
+for segment in segments:
+    print("[%.2fs -> %.2fs] %s" % (segment.start, segment.end, segment.text))

faster_whisper/__init__.py

+from faster_whisper.audio import decode_audio
+from faster_whisper.transcribe import BatchedInferencePipeline, WhisperModel
+from faster_whisper.utils import available_models, download_model, format_timestamp
+from faster_whisper.version import __version__
+
+__all__ = [
+    "available_models",
+    "decode_audio",
+    "WhisperModel",
+    "BatchedInferencePipeline",
+    "download_model",
+    "format_timestamp",
+    "__version__",
+]

faster_whisper/audio.py

+from typing import BinaryIO, Union
+
+import torch
+import torchaudio
+
+
+def decode_audio(
+    input_file: Union[str, BinaryIO],
+    sampling_rate: int = 16000,
+    split_stereo: bool = False,
+):
+    """Decodes the audio.
+
+    Args:
+      input_file: Path to the input file or a file-like object.
+      sampling_rate: Resample the audio to this sample rate.
+      split_stereo: Return separate left and right channels.
+
+    Returns:
+      A float32 Torch Tensor.
+
+      If `split_stereo` is enabled, the function returns a 2-tuple with the
+      separated left and right channels.
+    """
+
+    waveform, audio_sf = torchaudio.load(input_file)  # waveform: channels X T
+
+    if audio_sf != sampling_rate:
+        waveform = torchaudio.functional.resample(
+            waveform, orig_freq=audio_sf, new_freq=sampling_rate
+        )
+    if split_stereo:
+        return waveform[0], waveform[1]
+
+    return waveform.mean(0)
+
+
+def pad_or_trim(array, length: int, *, axis: int = -1):
+    """
+    Pad or trim the audio array to N_SAMPLES, as expected by the encoder.
+    """
+    axis = axis % array.ndim
+    if array.shape[axis] > length:
+        idx = [Ellipsis] * axis + [slice(length)] + [Ellipsis] * (array.ndim - axis - 1)
+        return array[idx]
+
+    if array.shape[axis] < length:
+        pad_widths = (
+            [
+                0,
+            ]
+            * array.ndim
+            * 2
+        )
+        pad_widths[2 * axis] = length - array.shape[axis]
+        array = torch.nn.functional.pad(array, tuple(pad_widths[::-1]))
+
+    return array