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	---
	license: mit
	language: fr
	library_name: transformers
	pipeline_tag: automatic-speech-recognition
	thumbnail: null
	tags:
	- automatic-speech-recognition
	- hf-asr-leaderboard
	datasets:
	- mozilla-foundation/common_voice_13_0
	- facebook/multilingual_librispeech
	- facebook/voxpopuli
	- google/fleurs
	- gigant/african_accented_french
	metrics:
	- wer
	model-index:
	- name: whisper-large-v3-french
	results:
	- task:
	name: Automatic Speech Recognition
	type: automatic-speech-recognition
	dataset:
	name: Common Voice 13.0
	type: mozilla-foundation/common_voice_13_0
	config: fr
	split: test
	args:
	language: fr
	metrics:
	- name: WER
	type: wer
	value: 7.28
	- task:
	name: Automatic Speech Recognition
	type: automatic-speech-recognition
	dataset:
	name: Multilingual LibriSpeech (MLS)
	type: facebook/multilingual_librispeech
	config: french
	split: test
	args:
	language: fr
	metrics:
	- name: WER
	type: wer
	value: 3.98
	- task:
	name: Automatic Speech Recognition
	type: automatic-speech-recognition
	dataset:
	name: VoxPopuli
	type: facebook/voxpopuli
	config: fr
	split: test
	args:
	language: fr
	metrics:
	- name: WER
	type: wer
	value: 8.91
	- task:
	name: Automatic Speech Recognition
	type: automatic-speech-recognition
	dataset:
	name: Fleurs
	type: google/fleurs
	config: fr_fr
	split: test
	args:
	language: fr
	metrics:
	- name: WER
	type: wer
	value: 4.84
	- task:
	name: Automatic Speech Recognition
	type: automatic-speech-recognition
	dataset:
	name: African Accented French
	type: gigant/african_accented_french
	config: fr
	split: test
	args:
	language: fr
	metrics:
	- name: WER
	type: wer
	value: 4.20
	---

	# Whisper-Large-V3-French

	Whisper-Large-V3-French is fine-tuned on `openai/whisper-large-v3` to further enhance its performance on the French language. This model has been trained to predict casing, punctuation, and numbers. While this might slightly sacrifice performance, we believe it allows for broader usage.

	This model has been converted into various formats, facilitating its usage across different libraries, including transformers, openai-whisper, fasterwhisper, whisper.cpp, candle, mlx, etc.

	## Table of Contents

	- [Performance](#performance)
	- [Usage](#usage)
	- [Hugging Face Pipeline](#hugging-face-pipeline)
	- [Hugging Face Low-level APIs](#hugging-face-low-level-apis)
	- [Speculative Decoding](#speculative-decoding)
	- [OpenAI Whisper](#openai-whisper)
	- [Faster Whisper](#faster-whisper)
	- [Whisper.cpp](#whispercpp)
	- [Candle](#candle)
	- [MLX](#mlx)
	- [Training details](#training-details)
	- [Acknowledgements](#acknowledgements)

	## Performance

	We evaluated our model on both short and long-form transcriptions, and also tested it on both in-distribution and out-of-distribution datasets to conduct a comprehensive analysis assessing its accuracy, generalizability, and robustness.

	Please note that the reported WER is the result after converting numbers to text, removing punctuation (except for apostrophes and hyphens), and converting all characters to lowercase.

	All evaluation results on the public datasets can be found [here](https://drive.google.com/drive/folders/1rFIh6yXRVa9RZ0ieZoKiThFZgQ4STPPI?usp=drive_link).

	### Short-Form Transcription

	![eval-short-form](https://huggingface.co/bofenghuang/whisper-large-v3-french/resolve/main/assets/whisper_fr_eval_short_form.png)

	Due to the lack of readily available out-of-domain (OOD) and long-form test sets in French, we evaluated using internal test sets from [Zaion Lab](https://zaion.ai/). These sets comprise human-annotated audio-transcription pairs from call center conversations, which are notable for their significant background noise and domain-specific terminology.

	### Long-Form Transcription

	![eval-long-form](https://huggingface.co/bofenghuang/whisper-large-v3-french/resolve/main/assets/whisper_fr_eval_long_form.png)

	The long-form transcription was run using the 🤗 Hugging Face pipeline for quicker evaluation. Audio files were segmented into 30-second chunks and processed in parallel.

	## Usage

	### Hugging Face Pipeline

	The model can easily used with the 🤗 Hugging Face [`pipeline`](https://huggingface.co/docs/transformers/main_classes/pipelines#transformers.AutomaticSpeechRecognitionPipeline) class for audio transcription.

	For long-form transcription (> 30 seconds), you can activate the process by passing the `chunk_length_s` argument. This approach segments the audio into smaller segments, processes them in parallel, and then joins them at the strides by finding the longest common sequence. While this chunked long-form approach may have a slight compromise in performance compared to OpenAI's sequential algorithm, it provides 9x faster inference speed.

	```python
	import torch
	from datasets import load_dataset
	from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline

	device = "cuda:0" if torch.cuda.is_available() else "cpu"
	torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32

	# Load model
	model_name_or_path = "bofenghuang/whisper-large-v3-french"
	processor = AutoProcessor.from_pretrained(model_name_or_path)
	model = AutoModelForSpeechSeq2Seq.from_pretrained(
	model_name_or_path,
	torch_dtype=torch_dtype,
	low_cpu_mem_usage=True,
	)
	model.to(device)

	# Init pipeline
	pipe = pipeline(
	"automatic-speech-recognition",
	model=model,
	feature_extractor=processor.feature_extractor,
	tokenizer=processor.tokenizer,
	torch_dtype=torch_dtype,
	device=device,
	# chunk_length_s=30, # for long-form transcription
	max_new_tokens=128,
	)

	# Example audio
	dataset = load_dataset("bofenghuang/asr-dummy", "fr", split="test")
	sample = dataset[0]["audio"]

	# Run pipeline
	result = pipe(sample)
	print(result["text"])
	```

	### Hugging Face Low-level APIs

	You can also use the 🤗 Hugging Face low-level APIs for transcription, offering greater control over the process, as demonstrated below:

	```python
	import torch
	from datasets import load_dataset
	from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor

	device = "cuda:0" if torch.cuda.is_available() else "cpu"
	torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32

	# Load model
	model_name_or_path = "bofenghuang/whisper-large-v3-french"
	processor = AutoProcessor.from_pretrained(model_name_or_path)
	model = AutoModelForSpeechSeq2Seq.from_pretrained(
	model_name_or_path,
	torch_dtype=torch_dtype,
	low_cpu_mem_usage=True,
	)
	model.to(device)

	# Example audio
	dataset = load_dataset("bofenghuang/asr-dummy", "fr", split="test")
	sample = dataset[0]["audio"]

	# Extract feautres
	input_features = processor(
	sample["array"], sampling_rate=sample["sampling_rate"], return_tensors="pt"
	).input_features


	# Generate tokens
	predicted_ids = model.generate(
	input_features.to(dtype=torch_dtype).to(device), max_new_tokens=128
	)

	# Detokenize to text
	transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True)[0]
	print(transcription)
	```

	### Speculative Decoding

	[Speculative decoding](https://huggingface.co/blog/whisper-speculative-decoding) can be achieved using a draft model, essentially a distilled version of Whisper. This approach guarantees identical outputs to using the main Whisper model alone, offers a 2x faster inference speed, and incurs only a slight increase in memory overhead.

	Since the distilled Whisper has the same encoder as the original, only its decoder need to be loaded, and encoder outputs are shared between the main and draft models during inference.

	Using speculative decoding with the Hugging Face pipeline is simple - just specify the `assistant_model` within the generation configurations.

	```python
	import torch
	from datasets import load_dataset
	from transformers import (
	AutoModelForCausalLM,
	AutoModelForSpeechSeq2Seq,
	AutoProcessor,
	pipeline,
	)

	device = "cuda:0" if torch.cuda.is_available() else "cpu"
	torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32

	# Load model
	model_name_or_path = "bofenghuang/whisper-large-v3-french"
	processor = AutoProcessor.from_pretrained(model_name_or_path)
	model = AutoModelForSpeechSeq2Seq.from_pretrained(
	model_name_or_path,
	torch_dtype=torch_dtype,
	low_cpu_mem_usage=True,
	)
	model.to(device)

	# Load draft model
	assistant_model_name_or_path = "bofenghuang/whisper-large-v3-french-distil-dec2"
	assistant_model = AutoModelForCausalLM.from_pretrained(
	assistant_model_name_or_path,
	torch_dtype=torch_dtype,
	low_cpu_mem_usage=True,
	)
	assistant_model.to(device)

	# Init pipeline
	pipe = pipeline(
	"automatic-speech-recognition",
	model=model,
	feature_extractor=processor.feature_extractor,
	tokenizer=processor.tokenizer,
	torch_dtype=torch_dtype,
	device=device,
	generate_kwargs={"assistant_model": assistant_model},
	max_new_tokens=128,
	)

	# Example audio
	dataset = load_dataset("bofenghuang/asr-dummy", "fr", split="test")
	sample = dataset[0]["audio"]

	# Run pipeline
	result = pipe(sample)
	print(result["text"])
	```

	### OpenAI Whisper

	You can also employ the sequential long-form decoding algorithm with a sliding window and temperature fallback, as outlined by OpenAI in their original [paper](https://arxiv.org/abs/2212.04356).

	First, install the [openai-whisper](https://github.com/openai/whisper) package:

	```bash
	pip install -U openai-whisper
	```

	Then, download the converted model:

	```bash
	python -c "from huggingface_hub import hf_hub_download; hf_hub_download(repo_id='bofenghuang/whisper-large-v3-french', filename='original_model.pt', local_dir='./models/whisper-large-v3-french')"
	```

	Now, you can transcirbe audio files by following the usage instructions provided in the repository:

	```python
	import whisper
	from datasets import load_dataset

	# Load model
	model = whisper.load_model("./models/whisper-large-v3-french/original_model.pt")

	# Example audio
	dataset = load_dataset("bofenghuang/asr-dummy", "fr", split="test")
	sample = dataset[0]["audio"]["array"].astype("float32")

	# Transcribe
	result = model.transcribe(sample, language="fr")
	print(result["text"])
	```

	### Faster Whisper

	Faster Whisper is a reimplementation of OpenAI's Whisper models and the sequential long-form decoding algorithm in the [CTranslate2](https://github.com/OpenNMT/CTranslate2) format.

	Compared to openai-whisper, it offers up to 4x faster inference speed, while consuming less memory. Additionally, the model can be quantized into int8, further enhancing its efficiency on both CPU and GPU.

	First, install the [faster-whisper](https://github.com/SYSTRAN/faster-whisper) package:

	```bash
	pip install faster-whisper
	```

	Then, download the model converted to the CTranslate2 format:

	```bash
	python -c "from huggingface_hub import snapshot_download; snapshot_download(repo_id='bofenghuang/whisper-large-v3-french', local_dir='./models/whisper-large-v3-french', allow_patterns='ctranslate2/*')"
	```

	Now, you can transcirbe audio files by following the usage instructions provided in the repository:

	```python
	from datasets import load_dataset
	from faster_whisper import WhisperModel

	# Load model
	model = WhisperModel("./models/whisper-large-v3-french/ctranslate2", device="cuda", compute_type="float16") # Run on GPU with FP16

	# Example audio
	dataset = load_dataset("bofenghuang/asr-dummy", "fr", split="test")
	sample = dataset[0]["audio"]["array"].astype("float32")

	segments, info = model.transcribe(sample, beam_size=5, language="fr")

	for segment in segments:
	print("[%.2fs -> %.2fs] %s" % (segment.start, segment.end, segment.text))
	```

	### Whisper.cpp

	Whisper.cpp is a reimplementation of OpenAI's Whisper models, crafted in plain C/C++ without any dependencies. It offers compatibility with various backends and platforms.

	Additionally, the model can be quantized to either 4-bit or 5-bit integers, further enhancing its efficiency.

	First, clone and build the [whisper.cpp](https://github.com/ggerganov/whisper.cpp) repository:

	```bash
	git clone https://github.com/ggerganov/whisper.cpp.git
	cd whisper.cpp

	# build the main example
	make
	```

	Next, download the converted ggml weights from the Hugging Face Hub:

	```bash
	# Download model quantized with Q5_0 method
	python -c "from huggingface_hub import hf_hub_download; hf_hub_download(repo_id='bofenghuang/whisper-large-v3-french', filename='ggml-model-q5_0.bin', local_dir='./models/whisper-large-v3-french')"
	```

	Now, you can transcribe an audio file using the following command:

	```bash
	./main -m ./models/whisper-large-v3-french/ggml-model-q5_0.bin -l fr -f /path/to/audio/file --print-colors
	```

	### Candle

	[Candle-whisper](https://github.com/huggingface/candle/tree/main/candle-examples/examples/whisper) is a reimplementation of OpenAI's Whisper models in the candle format - a lightweight ML framework built in Rust.

	First, clone the [candle](https://github.com/huggingface/candle) repository:

	```bash
	git clone https://github.com/huggingface/candle.git
	cd candle/candle-examples/examples/whisper
	```

	Transcribe an audio file using the following command:

	```bash
	cargo run --example whisper --release -- --model large-v3 --model-id bofenghuang/whisper-large-v3-french --language fr --input /path/to/audio/file
	```

	In order to use CUDA add `--features cuda` to the example command line:

	```bash
	cargo run --example whisper --release --features cuda -- --model large-v3 --model-id bofenghuang/whisper-large-v3-french --language fr --input /path/to/audio/file
	```

	### MLX

	[MLX-Whisper](https://github.com/ml-explore/mlx-examples/tree/main/whisper) is a reimplementation of OpenAI's Whisper models in the [MLX](https://github.com/ml-explore/mlx) format - a ML framework on Apple silicon. It supports features like lazy computation, unified memory management, etc.

	First, clone the [MLX Examples](https://github.com/ml-explore/mlx-examples) repository:

	```bash
	git clone https://github.com/ml-explore/mlx-examples.git
	cd mlx-examples/whisper
	```

	Next, install the dependencies:

	```bash
	pip install -r requirements.txt
	```

	Download the pytorch checkpoint in the original OpenAI format and convert it into MLX format (We haven't included the converted version here since the repository is already heavy and the conversion is very fast):

	```bash
	# Download
	python -c "from huggingface_hub import hf_hub_download; hf_hub_download(repo_id='bofenghuang/whisper-large-v3-french', filename='original_model.pt', local_dir='./models/whisper-large-v3-french')"
	# Convert into .npz
	python convert.py --torch-name-or-path ./models/whisper-large-v3-french/original_model.pt --mlx-path ./mlx_models/whisper-large-v3-french
	```

	Now, you can transcribe audio with:

	```python
	import whisper

	result = whisper.transcribe("/path/to/audio/file", path_or_hf_repo="mlx_models/whisper-large-v3-french", language="fr")
	print(result["text"])
	```

	## Training details

	We've collected a composite dataset consisting of over 2,500 hours of French speech recognition data, which incldues datasets such as [Common Voice 13.0](https://huggingface.co/datasets/mozilla-foundation/common_voice_13_0), [Multilingual LibriSpeech](https://huggingface.co/datasets/facebook/multilingual_librispeech), [Voxpopuli](https://huggingface.co/datasets/facebook/voxpopuli), [Fleurs](https://huggingface.co/datasets/google/fleurs), [Multilingual TEDx](https://www.openslr.org/100/), [MediaSpeech](https://www.openslr.org/108/), [African Accented French](https://huggingface.co/datasets/gigant/african_accented_french), etc.

	Given that some datasets, like MLS, only offer text without case or punctuation, we employed a customized version of 🤗 [Speechbox](https://github.com/huggingface/speechbox) to restore case and punctuation from a limited set of symbols using the [bofenghuang/whisper-large-v2-cv11-french](bofenghuang/whisper-large-v2-cv11-french) model.

	However, even within these datasets, we observed certain quality issues. These ranged from mismatches between audio and transcription in terms of language or content, poorly segmented utterances, to missing words in scripted speech, etc. We've built a pipeline to filter out many of these problematic utterances, aiming to enhance the dataset's quality. As a result, we excluded more than 10% of the data, and when we retrained the model, we noticed a significant reduction of hallucination.

	For training, we employed the [script](https://github.com/huggingface/transformers/blob/main/examples/pytorch/speech-recognition/run_speech_recognition_seq2seq.py) available in the 🤗 Transformers repository. The model training took place on the [Jean-Zay supercomputer](http://www.idris.fr/eng/jean-zay/jean-zay-presentation-eng.html) at GENCI, and we extend our gratitude to the IDRIS team for their responsive support throughout the project.

	## Acknowledgements

	- OpenAI for creating and open-sourcing the [Whisper model](https://arxiv.org/abs/2212.04356)
	- 🤗 Hugging Face for integrating the Whisper model and providing the training codebase within the [Transformers](https://github.com/huggingface/transformers) repository
	- [Genci](https://genci.fr/) for their generous contribution of GPU hours to this project