CohereLabs_cohere-transcribe-03-2026

license: apache-2.0 language:

• ar
• de
• el
• en
• es
• fr
• it
• ja
• ko
• nl
• pl
• pt
• vi
• zh pipeline_tag: automatic-speech-recognition tags:
• audio
• hf-asr-leaderboard
• speech-recognition
• transcription library_name: transformers

Cohere Transcribe

Cohere Transcribe is an open source release of a 2B parameter dedicated audio-in, text-out automatic speech recognition (ASR) model. The model supports 14 languages.

Developed by: Cohere and Cohere Labs. Point of Contact: Cohere Labs.

✨Try the Cohere Transcribe demo✨

Usage

Cohere Transcribe is supported natively in transformers. This is the recommended way to use the model for offline inference. For online inference, see the vLLM integration example below.

pip install transformers>=5.4.0 torch huggingface_hub soundfile librosa sentencepiece protobuf accelerate
pip install datasets  # only needed for long-form and non-English examples

Testing was carried out with torch==2.10.0 but it is expected to work with other versions.

Quick Start 🤗

Transcribe any audio file in a few lines:

from transformers import AutoProcessor, CohereAsrForConditionalGeneration
from transformers.audio_utils import load_audio
from huggingface_hub import hf_hub_download

processor = AutoProcessor.from_pretrained("CohereLabs/cohere-transcribe-03-2026")
model = CohereAsrForConditionalGeneration.from_pretrained("CohereLabs/cohere-transcribe-03-2026", device_map="auto")

audio_file = hf_hub_download(
    repo_id="CohereLabs/cohere-transcribe-03-2026",
    filename="demo/voxpopuli_test_en_demo.wav",
)
audio = load_audio(audio_file, sampling_rate=16000)

inputs = processor(audio, sampling_rate=16000, return_tensors="pt", language="en")
inputs.to(model.device, dtype=model.dtype)

outputs = model.generate(**inputs, max_new_tokens=256)
text = processor.decode(outputs, skip_special_tokens=True)
print(text)

from transformers import AutoProcessor, CohereAsrForConditionalGeneration
from datasets import load_dataset
import time

processor = AutoProcessor.from_pretrained("CohereLabs/cohere-transcribe-03-2026")
model = CohereAsrForConditionalGeneration.from_pretrained("CohereLabs/cohere-transcribe-03-2026", device_map="auto")

ds = load_dataset("distil-whisper/earnings22", "full", split="test", streaming=True)
sample = next(iter(ds))

audio_array = sample["audio"]["array"]
sr = sample["audio"]["sampling_rate"]
duration_s = len(audio_array) / sr
print(f"Audio duration: {duration_s / 60:.1f} minutes")

inputs = processor(audio=audio_array, sampling_rate=sr, return_tensors="pt", language="en")
audio_chunk_index = inputs.get("audio_chunk_index")
inputs.to(model.device, dtype=model.dtype)

start = time.time()
outputs = model.generate(**inputs, max_new_tokens=256)
text = processor.decode(outputs, skip_special_tokens=True, audio_chunk_index=audio_chunk_index, language="en")[0]
elapsed = time.time() - start
rtfx = duration_s / elapsed
print(f"Transcribed in {elapsed:.1f}s — RTFx: {rtfx:.1f}")
print(f"Transcription ({len(text.split())} words):")
print(text[:500] + "...")

inputs_pnc = processor(audio, sampling_rate=16000, return_tensors="pt", language="en", punctuation=True)
inputs_nopnc = processor(audio, sampling_rate=16000, return_tensors="pt", language="en", punctuation=False)

from transformers import AutoProcessor, CohereAsrForConditionalGeneration
from transformers.audio_utils import load_audio

processor = AutoProcessor.from_pretrained("CohereLabs/cohere-transcribe-03-2026")
model = CohereAsrForConditionalGeneration.from_pretrained("CohereLabs/cohere-transcribe-03-2026", device_map="auto")

audio_short = load_audio(
    "https://huggingface.co/datasets/hf-internal-testing/dummy-audio-samples/resolve/main/bcn_weather.mp3",
    sampling_rate=16000,
)
audio_long = load_audio(
    "https://huggingface.co/datasets/hf-internal-testing/dummy-audio-samples/resolve/main/obama_first_45_secs.mp3",
    sampling_rate=16000,
)

inputs = processor([audio_short, audio_long], sampling_rate=16000, return_tensors="pt", language="en")
audio_chunk_index = inputs.get("audio_chunk_index")
inputs.to(model.device, dtype=model.dtype)

outputs = model.generate(**inputs, max_new_tokens=256)
text = processor.decode(
    outputs, skip_special_tokens=True, audio_chunk_index=audio_chunk_index, language="en"
)
print(text)

from transformers import AutoProcessor, CohereAsrForConditionalGeneration
from datasets import load_dataset

processor = AutoProcessor.from_pretrained("CohereLabs/cohere-transcribe-03-2026")
model = CohereAsrForConditionalGeneration.from_pretrained("CohereLabs/cohere-transcribe-03-2026", device_map="auto")

ds = load_dataset("google/fleurs", "ja_jp", split="test", streaming=True)
ds_iter = iter(ds)
samples = [next(ds_iter) for _ in range(3)]

for sample in samples:
    audio = sample["audio"]["array"]
    sr = sample["audio"]["sampling_rate"]

    inputs = processor(audio, sampling_rate=sr, return_tensors="pt", language="ja")
    inputs.to(model.device, dtype=model.dtype)

    outputs = model.generate(**inputs, max_new_tokens=256)
    text = processor.decode(outputs, skip_special_tokens=True)
    print(f"REF: {sample['transcription']}\nHYP: {text}\n")

vLLM Integration

For production serving we recommend running via vLLM following the instructions below.

uv venv --python 3.12 --seed
source .venv/bin/activate

uv pip install -U vllm==0.19.0 --torch-backend=auto
uv pip install vllm[audio]
uv pip install librosa

vllm serve CohereLabs/cohere-transcribe-03-2026 --trust-remote-code

curl -v -X POST http://localhost:8000/v1/audio/transcriptions \
 -H "Authorization: Bearer $VLLM_API_KEY" \
-F "file=@$(realpath ${AUDIO_PATH})" \
-F "model=CohereLabs/cohere-transcribe-03-2026"

Results

Human-preference results

We observe similarly strong performance in human evaluations, where trained annotators assess transcription quality across real-world audio for accuracy, coherence and usability. The consistency between automated metrics and human judgments suggests that the model’s improvements translate beyond controlled benchmarks to practical transcription settings.

Figure: Human preference evaluation of model transcripts. In a head-to-head comparison, annotators were asked to express preferences for generations which primarily preserved meaning - but also avoided hallucination, correctly identified named entities, and provided verbatim transcripts with appropriate formatting. A score of 50% or higher indicates that Cohere Transcribe was preferred on average in the comparison.

Resources

For more details and results:

• Technical blog post contains WERs and other quality metrics.
• Announcement blog post for more information about the model.
• English, EU and long-form transcription WERs/RTFx are on the Open ASR Leaderboard.

Strengths and Limitations

Cohere Transcribe is a performant, dedicated ASR model intended for efficient speech transcription.

Strengths

Cohere Transcribe demonstrates best-in-class transcription accuracy in 14 languages. As a dedicated speech recognition model, it is also efficient, benefitting from a real-time factor up to three times faster than that of other, dedicated ASR models in the same size range. The model was trained from scratch, and from the outset, we deliberately focused on maximizing transcription accuracy while keeping production readiness top-of-mind.

Limitations

• Single language. The model performs best when remaining in-distribution of a single, pre-specified language amongst the 14 in the range it supports. It does not feature explicit, automatic language detection and exhibits inconsistent performance on code-switched audio.

• Timestamps/Speaker diarization. The model does not feature either of these.

• Silence. Like most AED speech models, Cohere Transcribe is eager to transcribe, even non-speech sounds. The model thus benefits from prepending a noise gate or VAD (voice activity detection) model in order to prevent low-volume, floor noise from turning into hallucinations.

Ecosystem support 🚀

Cohere Transcribe is supported on the following libraries/platforms:

• transformers (see Quick Start above).
• vLLM (see vLLM integration above).
• mlx-audio for Apple Silicon.
• Rust implementation: cohere_transcribe_rs
• In the browser ✨demo✨ (via transformers.js and WebGPU)
• Chrome extension: cohere_transcribe_extension
• nano-cohere-transcribe - blazing fast, particularly for long-form audio. e.g. 18 mins audio transcribed in 2.36s
• Whisper Memos (iOS App).
• Whisperian (Android App).
• hyprwhspr (Linux app).

If you have added support for the model somewhere not included above please raise an issue/PR!

If you find issues with any of these please raise an issue with the respective library.

Model Card Contact

For errors or additional questions about details in this model card, contact [email protected] or raise an issue.

Terms of Use: We hope that the release of this model will make community-based research efforts more accessible, by releasing the weights of a highly performant 2 billion parameter model to researchers all over the world. This model is governed by an Apache 2.0 license.

Citation

To cite this model please use the following bibtex:

@misc{julian_mack_2026,
	author       = { Julian Mack and Ekagra Ranjan and Walter Beller-Morales and Bharat Venkitesh and Pierre Richemond },
	title        = { cohere-transcribe-03-2026 (Revision d96e814) },
	year         = 2026,
	url          = { https://huggingface.co/CohereLabs/cohere-transcribe-03-2026 },
	doi          = { 10.57967/hf/8653 },
	publisher    = { Hugging Face }
}