# Copyright (c) 2024 NVIDIA CORPORATION. # Licensed under the MIT license. import os import yaml import gradio as gr import librosa from pydub import AudioSegment import soundfile as sf import numpy as np import torch import laion_clap from inference_utils import prepare_tokenizer, prepare_model, inference from data import AudioTextDataProcessor def load_laionclap(): model = laion_clap.CLAP_Module(enable_fusion=True, amodel='HTSAT-tiny').cuda() model.load_ckpt(ckpt='630k-audioset-fusion-best.pt') model.eval() return model 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) def load_audio(file_path, target_sr=44100, duration=33.25, start=0.0): if file_path.endswith('.mp3'): audio = AudioSegment.from_file(file_path) if len(audio) > (start + duration) * 1000: audio = audio[start * 1000:(start + duration) * 1000] if audio.frame_rate != target_sr: audio = audio.set_frame_rate(target_sr) if audio.channels > 1: audio = audio.set_channels(1) data = np.array(audio.get_array_of_samples()) if audio.sample_width == 2: data = data.astype(np.float32) / np.iinfo(np.int16).max elif audio.sample_width == 4: data = data.astype(np.float32) / np.iinfo(np.int32).max else: raise ValueError("Unsupported bit depth: {}".format(audio.sample_width)) else: with sf.SoundFile(file_path) as audio: original_sr = audio.samplerate channels = audio.channels max_frames = int((start + duration) * original_sr) audio.seek(int(start * original_sr)) frames_to_read = min(max_frames, len(audio)) data = audio.read(frames_to_read) if data.max() > 1 or data.min() < -1: data = data / max(abs(data.max()), abs(data.min())) if original_sr != target_sr: if channels == 1: data = librosa.resample(data.flatten(), orig_sr=original_sr, target_sr=target_sr) else: data = librosa.resample(data.T, orig_sr=original_sr, target_sr=target_sr)[0] else: if channels != 1: data = data.T[0] if data.min() >= 0: data = 2 * data / abs(data.max()) - 1.0 else: data = data / max(abs(data.max()), abs(data.min())) return data @torch.no_grad() def compute_laionclap_text_audio_sim(audio_file, laionclap_model, outputs): try: data = load_audio(audio_file, target_sr=48000) except Exception as e: print(audio_file, 'unsuccessful due to', e) return [0.0] * len(outputs) audio_data = data.reshape(1, -1) audio_data_tensor = torch.from_numpy(int16_to_float32(float32_to_int16(audio_data))).float().cuda() audio_embed = laionclap_model.get_audio_embedding_from_data(x=audio_data_tensor, use_tensor=True) text_embed = laionclap_model.get_text_embedding(outputs, use_tensor=True) cos = torch.nn.CosineSimilarity(dim=1, eps=1e-6) cos_similarity = cos(audio_embed.repeat(text_embed.shape[0], 1), text_embed) return cos_similarity.squeeze().cpu().numpy() inference_kwargs = { "do_sample": True, "top_k": 50, "top_p": 0.95, "num_return_sequences": 10 } config = yaml.load(open('chat.yaml'), Loader=yaml.FullLoader) clap_config = config['clap_config'] model_config = config['model_config'] text_tokenizer = prepare_tokenizer(model_config) DataProcessor = AudioTextDataProcessor( data_root='./', clap_config=clap_config, tokenizer=text_tokenizer, max_tokens=512, ) laionclap_model = load_laionclap() model = prepare_model( model_config=model_config, clap_config=clap_config, checkpoint_path='chat.pt' ) def inference_item(name, prompt): item = { 'name': str(name), 'prefix': 'The task is dialog.', 'prompt': str(prompt) } processed_item = DataProcessor.process(item) outputs = inference( model, text_tokenizer, item, processed_item, inference_kwargs, ) laionclap_scores = compute_laionclap_text_audio_sim( item["name"], laionclap_model, outputs ) outputs_joint = [(output, score) for (output, score) in zip(outputs, laionclap_scores)] outputs_joint.sort(key=lambda x: -x[1]) return outputs_joint[0][0] with gr.Blocks(title="Audio Flamingo - Demo") as ui: gr.HTML( """