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README.md
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---
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license: apache-2.0
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---
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# Model
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## TL;DR
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CLAP is to audio what CLIP is to image. This is an improved CLAP checkpoint, specifically trained on general audio, music and speech.
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## Description
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CLAP (Contrastive Language-Audio Pretraining) is a neural network trained on a variety of (audio, text) pairs. It can be instructed in to predict the most relevant text snippet, given an audio, without directly optimizing for the task. The CLAP model uses a SWINTransformer to get audio features from a log-Mel spectrogram input, and a RoBERTa model to get text features. Both the text and audio features are then projected to a latent space with identical dimension. The dot product between the projected audio and text features is then used as a similar score.
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# Usage
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You can use this model for zero shot audio classification or extracting audio and/or textual features.
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# Uses
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## Perform zero-shot audio classification
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### Using `pipeline`
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```python
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from datasets import load_dataset
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from transformers import pipeline
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dataset = load_dataset("ashraq/esc50")
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audio = dataset["train"]["audio"][-1]["array"]
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audio_classifier = pipeline(task="zero-shot-audio-classification", model="ylacombe/larger_clap_general")
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output = audio_classifier(audio, candidate_labels=["Sound of a dog", "Sound of vaccum cleaner"])
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print(output)
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>>> [{"score": 0.999, "label": "Sound of a dog"}, {"score": 0.001, "label": "Sound of vaccum cleaner"}]
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```
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## Run the model:
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You can also get the audio and text embeddings using `ClapModel`
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### Run the model on CPU:
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```python
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from datasets import load_dataset
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from transformers import ClapModel, ClapProcessor
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librispeech_dummy = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
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audio_sample = librispeech_dummy[0]
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model = ClapModel.from_pretrained("ylacombe/larger_clap_general")
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processor = ClapProcessor.from_pretrained("ylacombe/larger_clap_general")
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inputs = processor(audios=audio_sample["audio"]["array"], return_tensors="pt")
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audio_embed = model.get_audio_features(**inputs)
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```
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### Run the model on GPU:
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```python
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from datasets import load_dataset
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from transformers import ClapModel, ClapProcessor
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librispeech_dummy = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
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audio_sample = librispeech_dummy[0]
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model = ClapModel.from_pretrained("ylacombe/larger_clap_general").to(0)
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processor = ClapProcessor.from_pretrained("ylacombe/larger_clap_general")
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inputs = processor(audios=audio_sample["audio"]["array"], return_tensors="pt").to(0)
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audio_embed = model.get_audio_features(**inputs)
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```
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# Citation
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If you are using this model for your work, please consider citing the original paper:
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```
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@misc{https://doi.org/10.48550/arxiv.2211.06687,
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doi = {10.48550/ARXIV.2211.06687},
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url = {https://arxiv.org/abs/2211.06687},
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author = {Wu, Yusong and Chen, Ke and Zhang, Tianyu and Hui, Yuchen and Berg-Kirkpatrick, Taylor and Dubnov, Shlomo},
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keywords = {Sound (cs.SD), Audio and Speech Processing (eess.AS), FOS: Computer and information sciences, FOS: Computer and information sciences, FOS: Electrical engineering, electronic engineering, information engineering, FOS: Electrical engineering, electronic engineering, information engineering},
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title = {Large-scale Contrastive Language-Audio Pretraining with Feature Fusion and Keyword-to-Caption Augmentation},
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publisher = {arXiv},
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year = {2022},
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copyright = {Creative Commons Attribution 4.0 International}
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}
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```
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