Instructions to use CohereLabs/cohere-transcribe-arabic-07-2026 with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use CohereLabs/cohere-transcribe-arabic-07-2026 with Transformers:
# Use a pipeline as a high-level helper from transformers import pipeline pipe = pipeline("automatic-speech-recognition", model="CohereLabs/cohere-transcribe-arabic-07-2026")# Load model directly from transformers import AutoProcessor, AutoModelForSpeechSeq2Seq processor = AutoProcessor.from_pretrained("CohereLabs/cohere-transcribe-arabic-07-2026") model = AutoModelForSpeechSeq2Seq.from_pretrained("CohereLabs/cohere-transcribe-arabic-07-2026") - Notebooks
- Google Colab
- Kaggle
Cohere Transcribe Arabic
Cohere Transcribe Arabic is an open source release of a 2B parameter dedicated audio-in, text-out automatic speech recognition (ASR) model. The optimized for Arabic and English, with a focus on Arabic dialect performance and Arabic-English code-switching. Based on the Cohere Transcribe architecture.
Developed by: Cohere and Cohere Labs. Point of Contact: Cohere Labs.
| Name | cohere-transcribe-arabic-07-2026 |
|---|---|
| Architecture | conformer-based encoder-decoder |
| Input | audio waveform โ log-Mel spectrogram. Audio is automatically resampled to 16kHz if necessary during preprocessing. Similarly, multi-channel (stereo) inputs are averaged to produce a single channel signal. |
| Output | transcribed text |
| Model | a large Conformer encoder extracts acoustic representations, followed by a lightweight Transformer decoder for token generation |
| Training objective | supervised cross-entropy on output tokens |
| Languages |
|
| License | Apache 2.0 |
โจTry the Cohere Transcribe Arabic demoโจ
Usage
Cohere Transcribe Arabic 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
Quick Start ๐ค
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-arabic-07-2026")
model = CohereAsrForConditionalGeneration.from_pretrained("CohereLabs/cohere-transcribe-arabic-07-2026", device_map="auto")
# Example: transcribe Arabic audio
audio_file = "your_audio.wav"
audio = load_audio(audio_file, sampling_rate=16000)
inputs = processor(audio, sampling_rate=16000, return_tensors="pt", language="ar")
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)
Long-form transcription
For audio longer than the feature extractor's max_audio_clip_s, the feature extractor automatically splits the waveform into chunks.
The processor reassembles the per-chunk transcriptions using the returned audio_chunk_index.
from transformers import AutoProcessor, CohereAsrForConditionalGeneration
import time
processor = AutoProcessor.from_pretrained("CohereLabs/cohere-transcribe-arabic-07-2026")
model = CohereAsrForConditionalGeneration.from_pretrained("CohereLabs/cohere-transcribe-arabic-07-2026", device_map="auto")
audio = load_audio("your_long_audio.wav", sampling_rate=16000)
sr = 16000
duration_s = len(audio) / sr
print(f"Audio duration: {duration_s / 60:.1f} minutes")
inputs = processor(audio=audio, sampling_rate=sr, return_tensors="pt", language="ar")
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="ar")[0]
elapsed = time.time() - start
rtfx = duration_s / elapsed
print(f"Transcribed in {elapsed:.1f}s โ RTFx: {rtfx:.1f}")
print(text)
English transcription
The model also supports English. Specify language="en":
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)
vLLM Integration
For production serving we recommend running via vLLM following the instructions below.
Run cohere-transcribe-arabic-07-2026 via vLLM
First install vLLM (refer to vLLM installation instructions):
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
Start vLLM server
vllm serve CohereLabs/cohere-transcribe-arabic-07-2026 --trust-remote-code
Send request
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-arabic-07-2026"
Results
Open Universal Arabic ASR Leaderboard (as of 07.07.2026)
| Model | AverageWER ยท CER | SADAWER ยท CER | Common VoiceWER ยท CER | MASC cleanWER ยท CER | MASC noisyWER ยท CER | MGB-2WER ยท CER | CasablancaWER ยท CER |
|---|---|---|---|---|---|---|---|
| Cohere Transcribe Arabic 07-2026 | 25.8711.80 | 37.4723.53 | 5.821.62 | 19.606.45 | 27.0710.13 | 15.548.40 | 49.7120.66 |
| OmniASR LLM 7B | 28.3212.52 | 41.6124.95 | 8.752.71 | 19.695.76 | 29.2910.66 | 14.137.10 | 56.4623.96 |
| OmniASR LLM 3B | 29.9613.77 | 46.1827.27 | 9.152.80 | 19.906.13 | 30.0311.27 | 14.227.06 | 60.2728.06 |
| OmniASR LLM 1B | 29.9613.40 | 43.8424.54 | 9.552.97 | 20.036.14 | 30.2611.18 | 15.347.56 | 60.6828.02 |
| Cohere Transcribe 03-2026 | 30.6716.37 | 60.1145.44 | 8.172.49 | 8.662.97 | 19.017.71 | 25.339.28 | 62.7130.31 |
| Qwen3-Omni 30B | 30.7113.67 | 44.8226.11 | 11.464.28 | 21.475.59 | 30.8511.28 | 13.096.20 | 62.5528.53 |
| NVIDIA Conformer-CTC (LM) | 32.9113.84 | 44.5223.76 | 8.802.77 | 23.745.63 | 34.2911.07 | 17.206.87 | 68.9032.97 |
| OmniASR LLM 300M | 32.9614.84 | 51.3829.10 | 12.034.04 | 20.666.22 | 32.4512.23 | 16.587.86 | 64.6429.61 |
| Gemma 4 E4B | 32.9813.71 | 43.4020.96 | 19.657.48 | 24.867.76 | 33.5912.25 | 17.728.67 | 58.6325.11 |
| Qwen3-ASR 1.7B | 33.3612.33 | 45.5319.90 | 16.905.06 | 24.375.72 | 34.2910.84 | 16.576.25 | 64.4726.23 |
| Voxtral-Small 24B | 34.4715.29 | 50.8228.85 | 15.255.54 | 23.967.06 | 34.4312.22 | 16.037.41 | 66.3030.64 |
| NVIDIA Conformer-CTC (greedy) | 34.7413.37 | 47.2622.54 | 10.603.05 | 24.125.63 | 35.6411.02 | 19.697.46 | 71.1330.50 |
| Gemma 4 E2B | 35.8715.34 | 46.2323.47 | 23.769.13 | 27.478.99 | 36.1513.93 | 20.7210.15 | 60.8726.35 |
| Whisper Large v3 | 36.8617.21 | 55.9634.62 | 17.835.74 | 24.667.24 | 34.6312.89 | 16.267.74 | 71.8135.04 |
Link to the live leaderboard: Open Universal Arabic ASR Leaderboard.
Resources
For more details and results:
- Technical blog post contains WERs and other quality metrics.
- Announcement blog post for more information about the model.
- The Open Universal Arabic ASR Leaderboard.
Strengths and Limitations
Strengths
Cohere Transcribe Arabic demonstrates strong transcription accuracy for Arabic and English. As a dedicated speech recognition model, it benefits from efficient inference via the Conformer encoder-decoder architecture.
Limitations
Single language. The model performs best when remaining in-distribution of a single, pre-specified language. 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 Transcibe Arabic is eager to transcribe, even non-speech sounds. The model benefits from prepending a noise gate or VAD (voice activity detection) model in order to prevent low-volume, floor noise from turning into hallucinations.
Model Card Contact
For errors or additional questions about details in this model card, contact labs@cohere.com or raise an issue.
Terms of Use: We hope that the release of this model will make community-based research efforts into Arabic speech more accessible. This model is governed by an Apache 2.0 license.
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