import os import gradio as gr import torch import numpy as np import librosa from efficientat.models.MobileNetV3 import get_model as get_mobilenet, get_ensemble_model from efficientat.models.preprocess import AugmentMelSTFT from efficientat.helpers.utils import NAME_TO_WIDTH, labels from torch import autocast from contextlib import nullcontext from langchain import OpenAI, ConversationChain, LLMChain, PromptTemplate from langchain.chains.conversation.memory import ConversationalBufferWindowMemory MODEL_NAME = "mn40_as" device = torch.device('cuda') if torch.cuda.is_available() else torch.device('cpu') model = get_mobilenet(width_mult=NAME_TO_WIDTH(MODEL_NAME), pretrained_name=MODEL_NAME) model.to(device) model.eval() cached_audio_class = "c" template = None prompt = None chain = None formatted_classname = "tree" chain = None def format_classname(classname): return classname.capitalize() def audio_tag( audio_path, human_input, sample_rate=32000, window_size=800, hop_size=320, n_mels=128, cuda=True, ): (waveform, _) = librosa.core.load(audio_path, sr=sample_rate, mono=True) mel = AugmentMelSTFT(n_mels=n_mels, sr=sample_rate, win_length=window_size, hopsize=hop_size) mel.to(device) mel.eval() waveform = torch.from_numpy(waveform[None, :]).to(device) # our models are trained in half precision mode (torch.float16) # run on cuda with torch.float16 to get the best performance # running on cpu with torch.float32 gives similar performance, using torch.bfloat16 is worse with torch.no_grad(), autocast(device_type=device.type) if cuda and torch.cuda.is_available() else nullcontext(): spec = mel(waveform) preds, features = model(spec.unsqueeze(0)) preds = torch.sigmoid(preds.float()).squeeze().cpu().numpy() sorted_indexes = np.argsort(preds)[::-1] output = {} # Print audio tagging top probabilities label = labels[sorted_indexes[0]] formatted_classname = label chain = construct_langchain(formatted_classname) return formatted_classname def construct_langchain(audio_class): if cached_audio_class != audio_class: cached_audio_class = audio_class prefix = f"""You are going to act as a magical tool that allows for humans to communicate with non-human entities like rocks, crackling fire, trees, animals, and the wind. In order to do this, we're going to provide you the human's text input for the conversation. The goal is for you to embody that non-human entity and converse with the human. Examples: Non-human Entity: Tree Human Input: Hello tree Tree: Hello human, I am a tree Let's begin: Non-human Entity: {audio_class}""" suffix = f'''Source: {audio_class} Length of Audio in Seconds: 2 seconds Human Input: {userText} {audio_class} Response:''' template = prefix + suffix prompt = PromptTemplate( input_variables=["history", "human_input"], template=template ) chatgpt_chain = LLMChain( llm=OpenAI(temperature=.5, openai_api_key=session_token), prompt=prompt, verbose=True, memory=ConversationalBufferWindowMemory(k=2, ai_prefix=audio_class), ) return chatgpt_chain def predict(input, history=[]): formatted_message = chain.predict(human_input=input) history.append(formatted_message) return formatted_message, history demo = gr.Interface( fn=predict, [ gr.Audio(source="upload", type="filepath", label="Your audio"), ], inputs=["text", "state"], outputs=["chatbot", "state"], title="AnyChat", description="Non-Human entities have many things to say, listen to them!", ).launch(debug=True)