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| """ | |
| Contrastive Language-Audio Pretraining Model from LAION | |
| -------------------------------------------------------- | |
| Paper: https://arxiv.org/abs/2211.06687 | |
| Authors (equal contributions): Ke Chen, Yusong Wu, Tianyu Zhang, Yuchen Hui | |
| Support: LAION | |
| """ | |
| import numpy as np | |
| import librosa | |
| import torch | |
| import laion_clap | |
| # quantization | |
| def int16_to_float32(x): | |
| return (x / 32767.0).astype(np.float32) | |
| def float32_to_int16(x): | |
| x = np.clip(x, a_min=-1., a_max=1.) | |
| return (x * 32767.).astype(np.int16) | |
| model = laion_clap.CLAP_Module(enable_fusion=False) | |
| model.load_ckpt() | |
| # Directly get audio embeddings from audio files | |
| audio_file = [ | |
| '/home/la/kechen/Research/KE_CLAP/ckpt/test_clap_short.wav', | |
| '/home/la/kechen/Research/KE_CLAP/ckpt/test_clap_long.wav' | |
| ] | |
| audio_embed = model.get_audio_embedding_from_filelist(x = audio_file, use_tensor=False) | |
| print(audio_embed[:,-20:]) | |
| print(audio_embed.shape) | |
| # Get audio embeddings from audio data | |
| audio_data, _ = librosa.load('/home/la/kechen/Research/KE_CLAP/ckpt/test_clap_short.wav', sr=48000) # sample rate should be 48000 | |
| audio_data = audio_data.reshape(1, -1) # Make it (1,T) or (N,T) | |
| audio_embed = model.get_audio_embedding_from_data(x = audio_data, use_tensor=False) | |
| print(audio_embed[:,-20:]) | |
| print(audio_embed.shape) | |
| # Directly get audio embeddings from audio files, but return torch tensor | |
| audio_file = [ | |
| '/home/la/kechen/Research/KE_CLAP/ckpt/test_clap_short.wav', | |
| '/home/la/kechen/Research/KE_CLAP/ckpt/test_clap_long.wav' | |
| ] | |
| audio_embed = model.get_audio_embedding_from_filelist(x = audio_file, use_tensor=True) | |
| print(audio_embed[:,-20:]) | |
| print(audio_embed.shape) | |
| # Get audio embeddings from audio data | |
| audio_data, _ = librosa.load('/home/la/kechen/Research/KE_CLAP/ckpt/test_clap_short.wav', sr=48000) # sample rate should be 48000 | |
| audio_data = audio_data.reshape(1, -1) # Make it (1,T) or (N,T) | |
| audio_data = torch.from_numpy(int16_to_float32(float32_to_int16(audio_data))).float() # quantize before send it in to the model | |
| audio_embed = model.get_audio_embedding_from_data(x = audio_data, use_tensor=True) | |
| print(audio_embed[:,-20:]) | |
| print(audio_embed.shape) | |
| # Get text embedings from texts: | |
| text_data = ["I love the contrastive learning", "I love the pretrain model"] | |
| text_embed = model.get_text_embedding(text_data) | |
| print(text_embed) | |
| print(text_embed.shape) | |
| # Get text embedings from texts, but return torch tensor: | |
| text_data = ["I love the contrastive learning", "I love the pretrain model"] | |
| text_embed = model.get_text_embedding(text_data, use_tensor=True) | |
| print(text_embed) | |
| print(text_embed.shape) | |