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AudioGPT/audio-chatgpt.py

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import sys
import os
sys.path.append(os.path.dirname(os.path.realpath(__file__)))
sys.path.append(os.path.dirname(os.path.dirname(os.path.realpath(__file__))))
import gradio as gr
from transformers import AutoModelForCausalLM, AutoTokenizer, CLIPSegProcessor, CLIPSegForImageSegmentation
import torch
from diffusers import StableDiffusionPipeline
from diffusers import StableDiffusionInstructPix2PixPipeline, EulerAncestralDiscreteScheduler
import os
from langchain.agents.initialize import initialize_agent
from langchain.agents.tools import Tool
from langchain.chains.conversation.memory import ConversationBufferMemory
from langchain.llms.openai import OpenAI
import re
import uuid
import soundfile
from diffusers import StableDiffusionInpaintPipeline
from PIL import Image
import numpy as np
from omegaconf import OmegaConf
from transformers import pipeline, BlipProcessor, BlipForConditionalGeneration, BlipForQuestionAnswering
import cv2
import einops
from pytorch_lightning import seed_everything
import random
from ldm.util import instantiate_from_config
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from ldm.data.extract_mel_spectrogram import TRANSFORMS_16000
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from pathlib import Path
from vocoder.hifigan.modules import VocoderHifigan
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from Make_An_Audio_img.vocoder.bigvgan.models import VocoderBigVGAN
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from ldm.models.diffusion.ddim import DDIMSampler
from wav_evaluation.models.CLAPWrapper import CLAPWrapper
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from Make_An_Audio_img.ldm.util import instantiate_from_config as instantiate_from_config_make_an_audio_img
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import whisper
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AUDIO_CHATGPT_PREFIX = """Audio ChatGPT
TOOLS:
------
Audio ChatGPT has access to the following tools:"""
AUDIO_CHATGPT_FORMAT_INSTRUCTIONS = """To use a tool, please use the following format:
```
Thought: Do I need to use a tool? Yes
Action: the action to take, should be one of [{tool_names}]
Action Input: the input to the action
Observation: the result of the action
```
When you have a response to say to the Human, or if you do not need to use a tool, you MUST use the format:
```
Thought: Do I need to use a tool? No
{ai_prefix}: [your response here]
```
"""
AUDIO_CHATGPT_SUFFIX = """You are very strict to the filename correctness and will never fake a file name if not exists.
You will remember to provide the image file name loyally if it's provided in the last tool observation.
Begin!
Previous conversation history:
{chat_history}
New input: {input}
Thought: Do I need to use a tool? {agent_scratchpad}"""
SAMPLE_RATE = 16000
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temp_audio_filename = "audio/c00d9240.wav"
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def cut_dialogue_history(history_memory, keep_last_n_words = 500):
tokens = history_memory.split()
n_tokens = len(tokens)
print(f"hitory_memory:{history_memory}, n_tokens: {n_tokens}")
if n_tokens < keep_last_n_words:
return history_memory
else:
paragraphs = history_memory.split('\n')
last_n_tokens = n_tokens
while last_n_tokens >= keep_last_n_words:
last_n_tokens = last_n_tokens - len(paragraphs[0].split(' '))
paragraphs = paragraphs[1:]
return '\n' + '\n'.join(paragraphs)
def get_new_image_name(org_img_name, func_name="update"):
head_tail = os.path.split(org_img_name)
head = head_tail[0]
tail = head_tail[1]
name_split = tail.split('.')[0].split('_')
this_new_uuid = str(uuid.uuid4())[0:4]
if len(name_split) == 1:
most_org_file_name = name_split[0]
recent_prev_file_name = name_split[0]
new_file_name = '{}_{}_{}_{}.png'.format(this_new_uuid, func_name, recent_prev_file_name, most_org_file_name)
else:
assert len(name_split) == 4
most_org_file_name = name_split[3]
recent_prev_file_name = name_split[0]
new_file_name = '{}_{}_{}_{}.png'.format(this_new_uuid, func_name, recent_prev_file_name, most_org_file_name)
return os.path.join(head, new_file_name)
def initialize_model(config, ckpt, device):
config = OmegaConf.load(config)
model = instantiate_from_config(config.model)
model.load_state_dict(torch.load(ckpt,map_location='cpu')["state_dict"], strict=False)
model = model.to(device)
model.cond_stage_model.to(model.device)
model.cond_stage_model.device = model.device
sampler = DDIMSampler(model)
return sampler
clap_model = CLAPWrapper('useful_ckpts/CLAP/CLAP_weights_2022.pth','useful_ckpts/CLAP/config.yml',use_cuda=torch.cuda.is_available())
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def initialize_model_img(config, ckpt, device):
config = OmegaConf.load(config)
model = instantiate_from_config_make_an_audio_img(config.model)
model.load_state_dict(torch.load(ckpt,map_location='cpu')["state_dict"], strict=False)
model = model.to(device)
model.cond_stage_model.to(model.device)
model.cond_stage_model.device = model.device
sampler = DDIMSampler(model)
return sampler
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def select_best_audio(prompt,wav_list):
text_embeddings = clap_model.get_text_embeddings([prompt])
score_list = []
for data in wav_list:
sr,wav = data
audio_embeddings = clap_model.get_audio_embeddings([(torch.FloatTensor(wav),sr)], resample=True)
score = clap_model.compute_similarity(audio_embeddings, text_embeddings,use_logit_scale=False).squeeze().cpu().numpy()
score_list.append(score)
max_index = np.array(score_list).argmax()
print(score_list,max_index)
return wav_list[max_index]
class MaskFormer:
def __init__(self, device):
self.device = device
self.processor = CLIPSegProcessor.from_pretrained("CIDAS/clipseg-rd64-refined")
self.model = CLIPSegForImageSegmentation.from_pretrained("CIDAS/clipseg-rd64-refined").to(device)
def inference(self, image_path, text):
threshold = 0.5
min_area = 0.02
padding = 20
original_image = Image.open(image_path)
image = original_image.resize((512, 512))
inputs = self.processor(text=text, images=image, padding="max_length", return_tensors="pt",).to(self.device)
with torch.no_grad():
outputs = self.model(**inputs)
mask = torch.sigmoid(outputs[0]).squeeze().cpu().numpy() > threshold
area_ratio = len(np.argwhere(mask)) / (mask.shape[0] * mask.shape[1])
if area_ratio < min_area:
return None
true_indices = np.argwhere(mask)
mask_array = np.zeros_like(mask, dtype=bool)
for idx in true_indices:
padded_slice = tuple(slice(max(0, i - padding), i + padding + 1) for i in idx)
mask_array[padded_slice] = True
visual_mask = (mask_array * 255).astype(np.uint8)
image_mask = Image.fromarray(visual_mask)
return image_mask.resize(image.size)
class T2I:
def __init__(self, device):
print("Initializing T2I to %s" % device)
self.device = device
self.pipe = StableDiffusionPipeline.from_pretrained("runwayml/stable-diffusion-v1-5", torch_dtype=torch.float16)
self.text_refine_tokenizer = AutoTokenizer.from_pretrained("Gustavosta/MagicPrompt-Stable-Diffusion")
self.text_refine_model = AutoModelForCausalLM.from_pretrained("Gustavosta/MagicPrompt-Stable-Diffusion")
self.text_refine_gpt2_pipe = pipeline("text-generation", model=self.text_refine_model, tokenizer=self.text_refine_tokenizer, device=self.device)
self.pipe.to(device)
def inference(self, text):
image_filename = os.path.join('image', str(uuid.uuid4())[0:8] + ".png")
refined_text = self.text_refine_gpt2_pipe(text)[0]["generated_text"]
print(f'{text} refined to {refined_text}')
image = self.pipe(refined_text).images[0]
image.save(image_filename)
print(f"Processed T2I.run, text: {text}, image_filename: {image_filename}")
return image_filename
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class ImageCaptioning:
def __init__(self, device):
print("Initializing ImageCaptioning to %s" % device)
self.device = device
self.processor = BlipProcessor.from_pretrained("Salesforce/blip-image-captioning-base")
self.model = BlipForConditionalGeneration.from_pretrained("Salesforce/blip-image-captioning-base").to(self.device)
def inference(self, image_path):
inputs = self.processor(Image.open(image_path), return_tensors="pt").to(self.device)
out = self.model.generate(**inputs)
captions = self.processor.decode(out[0], skip_special_tokens=True)
return captions
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class T2A:
def __init__(self, device):
print("Initializing Make-An-Audio to %s" % device)
self.device = device
self.sampler = initialize_model('configs/text-to-audio/txt2audio_args.yaml', 'useful_ckpts/ta40multi_epoch=000085.ckpt', device=device)
self.vocoder = VocoderHifigan('vocoder/logs/hifi_0127',device=device)
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def txt2audio(self, text, seed = 55, scale = 1.5, ddim_steps = 100, n_samples = 3, W = 624, H = 80):
prng = np.random.RandomState(seed)
start_code = prng.randn(n_samples, self.sampler.model.first_stage_model.embed_dim, H // 8, W // 8)
start_code = torch.from_numpy(start_code).to(device=self.device, dtype=torch.float32)
uc = self.sampler.model.get_learned_conditioning(n_samples * [""])
c = self.sampler.model.get_learned_conditioning(n_samples * [text])
shape = [self.sampler.model.first_stage_model.embed_dim, H//8, W//8] # (z_dim, 80//2^x, 848//2^x)
samples_ddim, _ = self.sampler.sample(S = ddim_steps,
conditioning = c,
batch_size = n_samples,
shape = shape,
verbose = False,
unconditional_guidance_scale = scale,
unconditional_conditioning = uc,
x_T = start_code)
x_samples_ddim = self.sampler.model.decode_first_stage(samples_ddim)
x_samples_ddim = torch.clamp((x_samples_ddim+1.0)/2.0, min=0.0, max=1.0) # [0, 1]
wav_list = []
for idx,spec in enumerate(x_samples_ddim):
wav = self.vocoder.vocode(spec)
wav_list.append((SAMPLE_RATE,wav))
best_wav = select_best_audio(text, wav_list)
return best_wav
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def inference(self, text, seed = 55, scale = 1.5, ddim_steps = 100, n_samples = 3, W = 624, H = 80):
global temp_audio_filename
melbins,mel_len = 80,624
with torch.no_grad():
result = self.txt2audio(
text = text,
H = melbins,
W = mel_len
)
audio_filename = os.path.join('audio', str(uuid.uuid4())[0:8] + ".wav")
temp_audio_filename = audio_filename
soundfile.write(audio_filename, result[1], samplerate = 16000)
print(f"Processed T2I.run, text: {text}, audio_filename: {audio_filename}")
return audio_filename
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class I2A:
def __init__(self, device):
print("Initializing Make-An-Audio-Image to %s" % device)
self.device = device
self.sampler = initialize_model_img('Make_An_Audio_img/configs/img_to_audio/img2audio_args.yaml', 'Make_An_Audio_img/useful_ckpts/ta54_epoch=000216.ckpt', device=device)
self.vocoder = VocoderBigVGAN('Make_An_Audio_img/vocoder/logs/bigv16k53w',device=device)
def img2audio(self, image, seed = 55, scale = 3, ddim_steps = 100, W = 624, H = 80):
n_samples = 1 # only support 1 sample
prng = np.random.RandomState(seed)
start_code = prng.randn(n_samples, self.sampler.model.first_stage_model.embed_dim, H // 8, W // 8)
start_code = torch.from_numpy(start_code).to(device=self.device, dtype=torch.float32)
uc = self.sampler.model.get_learned_conditioning(n_samples * [""])
#image = Image.fromarray(image)
image = Image.open(image)
image = self.sampler.model.cond_stage_model.preprocess(image).unsqueeze(0)
image_embedding = self.sampler.model.cond_stage_model.forward_img(image)
c = image_embedding.repeat(n_samples, 1, 1)# shape:[1,77,1280],即还没有变成句子embedding仍是每个单词的embedding
shape = [self.sampler.model.first_stage_model.embed_dim, H//8, W//8] # (z_dim, 80//2^x, 848//2^x)
samples_ddim, _ = self.sampler.sample(S=ddim_steps,
conditioning=c,
batch_size=n_samples,
shape=shape,
verbose=False,
unconditional_guidance_scale=scale,
unconditional_conditioning=uc,
x_T=start_code)
x_samples_ddim = self.sampler.model.decode_first_stage(samples_ddim)
x_samples_ddim = torch.clamp((x_samples_ddim+1.0)/2.0, min=0.0, max=1.0) # [0, 1]
wav_list = []
for idx,spec in enumerate(x_samples_ddim):
wav = self.vocoder.vocode(spec)
wav_list.append((SAMPLE_RATE,wav))
best_wav = wav_list[0]
return best_wav
def inference(self, image, seed = 55, scale = 3, ddim_steps = 100, W = 624, H = 80):
global temp_audio_filename
melbins,mel_len = 80,624
with torch.no_grad():
result = self.img2audio(
image=image,
H=melbins,
W=mel_len
)
audio_filename = os.path.join('audio', str(uuid.uuid4())[0:8] + ".wav")
temp_audio_filename = audio_filename
soundfile.write(audio_filename, result[1], samplerate = 16000)
print(f"Processed I2a.run, image_filename: {image}, audio_filename: {audio_filename}")
return audio_filename
# need to debug
class Inpaint:
def __init__(self, device):
print("Initializing Make-An-Audio-inpaint to %s" % device)
self.device = device
self.sampler = initialize_model('Make_An_Audio_inpaint/configs/inpaint/txt2audio_args.yaml', 'Make_An_Audio_inpaint/useful_ckpts/inpaint7_epoch00047.ckpt')
self.vocoder = VocoderBigVGAN('./vocoder/logs/bigv16k53w',device=device)
def make_batch_sd(mel, mask, num_samples=1):
mel = torch.from_numpy(mel)[None,None,...].to(dtype=torch.float32)
mask = torch.from_numpy(mask)[None,None,...].to(dtype=torch.float32)
masked_mel = (1 - mask) * mel
mel = mel * 2 - 1
mask = mask * 2 - 1
masked_mel = masked_mel * 2 -1
batch = {
"mel": repeat(mel.to(device=self.device), "1 ... -> n ...", n=num_samples),
"mask": repeat(mask.to(device=self.device), "1 ... -> n ...", n=num_samples),
"masked_mel": repeat(masked_mel.to(device=self.device), "1 ... -> n ...", n=num_samples),
}
return batch
def gen_mel(input_audio):
sr,ori_wav = input_audio
print(sr,ori_wav.shape,ori_wav)
ori_wav = ori_wav.astype(np.float32, order='C') / 32768.0 # order='C'是以C语言格式存储不用管
if len(ori_wav.shape)==2:# stereo
ori_wav = librosa.to_mono(ori_wav.T)# gradio load wav shape could be (wav_len,2) but librosa expects (2,wav_len)
print(sr,ori_wav.shape,ori_wav)
ori_wav = librosa.resample(ori_wav,orig_sr = sr,target_sr = SAMPLE_RATE)
mel_len,hop_size = 848,256
input_len = mel_len * hop_size
if len(ori_wav) < input_len:
input_wav = np.pad(ori_wav,(0,mel_len*hop_size),constant_values=0)
else:
input_wav = ori_wav[:input_len]
mel = TRANSFORMS_16000(input_wav)
return mel
def show_mel_fn(input_audio):
crop_len = 500 # the full mel cannot be showed due to gradio's Image bug when using tool='sketch'
crop_mel = self.gen_mel(input_audio)[:,:crop_len]
color_mel = cmap_transform(crop_mel)
return Image.fromarray((color_mel*255).astype(np.uint8))
def inpaint(batch, seed, ddim_steps, num_samples=1, W=512, H=512):
model = self.sampler.model
prng = np.random.RandomState(seed)
start_code = prng.randn(num_samples, model.first_stage_model.embed_dim, H // 8, W // 8)
start_code = torch.from_numpy(start_code).to(device=self.device, dtype=torch.float32)
c = model.get_first_stage_encoding(model.encode_first_stage(batch["masked_mel"]))
cc = torch.nn.functional.interpolate(batch["mask"],
size=c.shape[-2:])
c = torch.cat((c, cc), dim=1) # (b,c+1,h,w) 1 is mask
shape = (c.shape[1]-1,)+c.shape[2:]
samples_ddim, _ = self.sampler.sample(S=ddim_steps,
conditioning=c,
batch_size=c.shape[0],
shape=shape,
verbose=False)
x_samples_ddim = model.decode_first_stage(samples_ddim)
mask = batch["mask"]# [-1,1]
mel = torch.clamp((batch["mel"]+1.0)/2.0,min=0.0, max=1.0)
mask = torch.clamp((batch["mask"]+1.0)/2.0,min=0.0, max=1.0)
predicted_mel = torch.clamp((x_samples_ddim+1.0)/2.0,min=0.0, max=1.0)
inpainted = (1-mask)*mel+mask*predicted_mel
inpainted = inpainted.cpu().numpy().squeeze()
inapint_wav = self.vocoder.vocode(inpainted)
return inpainted, inapint_wav
def predict(input_audio,mel_and_mask,ddim_steps,seed):
show_mel = np.array(mel_and_mask['image'].convert("L"))/255 # 由于展示的mel只展示了一部分所以需要重新从音频生成mel
mask = np.array(mel_and_mask["mask"].convert("L"))/255
mel_bins,mel_len = 80,848
input_mel = self.gen_mel(input_audio)[:,:mel_len]# 由于展示的mel只展示了一部分所以需要重新从音频生成mel
mask = np.pad(mask,((0,0),(0,mel_len-mask.shape[1])),mode='constant',constant_values=0)# 将mask填充到原来的mel的大小
print(mask.shape,input_mel.shape)
with torch.no_grad():
batch = make_batch_sd(input_mel,mask,device,num_samples=1)
inpainted,gen_wav = self.inpaint(
batch=batch,
seed=seed,
ddim_steps=ddim_steps,
num_samples=1,
H=mel_bins, W=mel_len
)
inpainted = inpainted[:,:show_mel.shape[1]]
color_mel = cmap_transform(inpainted)
input_len = int(input_audio[1].shape[0] * SAMPLE_RATE / input_audio[0])
gen_wav = (gen_wav * 32768).astype(np.int16)[:input_len]
return Image.fromarray((color_mel*255).astype(np.uint8)),(SAMPLE_RATE,gen_wav)
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class ASR:
def __init__(self, device):
print("Initializing Whisper to %s" % device)
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self.device = device
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self.model = whisper.load_model("base", device=device)
def inference(self, audio_path):
audio = whisper.load_audio(audio_path)
audio = whisper.pad_or_trim(audio)
mel = whisper.log_mel_spectrogram(audio).to(self.device)
_, probs = self.model.detect_language(mel)
#print(f"Detected language: {max(probs, key=probs.get)}")
options = whisper.DecodingOptions()
result = whisper.decode(self.model, mel, options)
return result.text
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class ConversationBot:
def __init__(self):
print("Initializing AudioChatGPT")
self.llm = OpenAI(temperature=0)
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self.t2i = T2I(device="cuda:1")
self.i2t = ImageCaptioning(device="cuda:1")
self.t2a = T2A(device="cuda:0")
self.i2a = I2A(device="cuda:1")
self.asr = ASR(device="cuda:1")
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self.memory = ConversationBufferMemory(memory_key="chat_history", output_key='output')
self.tools = [
Tool(name="Generate Image From User Input Text", func=self.t2i.inference,
description="useful for when you want to generate an image from a user input text and it saved it to a file. like: generate an image of an object or something, or generate an image that includes some objects. "
"The input to this tool should be a string, representing the text used to generate image. "),
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Tool(name="Get Photo Description", func=self.i2t.inference,
description="useful for when you want to know what is inside the photo. receives image_path as input. "
"The input to this tool should be a string, representing the image_path. "),
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Tool(name="Generate Audio From User Input Text", func=self.t2a.inference,
description="useful for when you want to generate an audio from a user input text and it saved it to a file."
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"The input to this tool should be a string, representing the text used to generate audio."),
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Tool(name="Generate Audio From The Image", func=self.i2a.inference,
description="useful for when you want to generate an audio based on an image."
"The input to this tool should be a string, representing the image_path. "),
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Tool(name="Get Audio Transcription", func=self.asr.inference,
description="useful for when you want to know the text content corresponding to this audio, receives audio_path as input."
"The input to this tool should be a string, representing the audio_path.")
]
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self.agent = initialize_agent(
self.tools,
self.llm,
agent="conversational-react-description",
verbose=True,
memory=self.memory,
return_intermediate_steps=True,
agent_kwargs={'prefix': AUDIO_CHATGPT_PREFIX, 'format_instructions': AUDIO_CHATGPT_FORMAT_INSTRUCTIONS, 'suffix': AUDIO_CHATGPT_SUFFIX}, )
def run_text(self, text, state):
print("===============Running run_text =============")
print("Inputs:", text, state)
print("======>Previous memory:\n %s" % self.agent.memory)
self.agent.memory.buffer = cut_dialogue_history(self.agent.memory.buffer, keep_last_n_words=500)
res = self.agent({"input": text})
print("======>Current memory:\n %s" % self.agent.memory)
response = re.sub('(image/\S*png)', lambda m: f'![](/file={m.group(0)})*{m.group(0)}*', res['output'])
state = state + [(text, response)]
print("Outputs:", state)
return state, state, temp_audio_filename
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def run_audio(self, audio, state, txt):
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#print(audio.type)
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print("===============Running run_audio =============")
print("Inputs:", audio, state)
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print("======>Previous memory:\n %s" % self.agent.memory)
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audio_filename = os.path.join('audio', str(uuid.uuid4())[0:8] + ".wav")
print("======>Auto Resize Audio...")
audio_load = whisper.load_audio(audio.name)
soundfile.write(audio_filename, audio_load, samplerate = 16000)
global temp_audio_filename
temp_audio_filename = audio_filename
description = self.asr.inference(audio_filename)
Human_prompt = "\nHuman: provide an audio named {}. The description is: {}. This information helps you to understand this audio, but you should use tools to finish following tasks, " \
"rather than directly imagine from my description. If you understand, say \"Received\". \n".format(audio_filename, description)
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AI_prompt = "Received. "
self.agent.memory.buffer = self.agent.memory.buffer + Human_prompt + 'AI: ' + AI_prompt
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state = state + [(f"*{audio_filename}*", AI_prompt)]
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print("Outputs:", state)
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return state, state, txt + ' ' + audio_filename + ' ', temp_audio_filename
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def run_image_or_audio(self, file, state, txt):
file_type = file.name[-3:]
if file_type == "wav":
print("===============Running run_audio =============")
print("Inputs:", file, state)
print("======>Previous memory:\n %s" % self.agent.memory)
audio_filename = os.path.join('audio', str(uuid.uuid4())[0:8] + ".wav")
print("======>Auto Resize Audio...")
audio_load = whisper.load_audio(file.name)
soundfile.write(audio_filename, audio_load, samplerate = 16000)
global temp_audio_filename
temp_audio_filename = audio_filename
description = self.asr.inference(audio_filename)
Human_prompt = "\nHuman: provide an audio named {}. The description is: {}. This information helps you to understand this audio, but you should use tools to finish following tasks, " \
"rather than directly imagine from my description. If you understand, say \"Received\". \n".format(audio_filename, description)
AI_prompt = "Received. "
self.agent.memory.buffer = self.agent.memory.buffer + Human_prompt + 'AI: ' + AI_prompt
state = state + [(f"*{audio_filename}*", AI_prompt)]
print("Outputs:", state)
return state, state, txt + ' ' + audio_filename + ' ', temp_audio_filename
else:
print("===============Running run_image =============")
print("Inputs:", file, state)
print("======>Previous memory:\n %s" % self.agent.memory)
image_filename = os.path.join('image', str(uuid.uuid4())[0:8] + ".png")
print("======>Auto Resize Image...")
img = Image.open(file.name)
width, height = img.size
ratio = min(512 / width, 512 / height)
width_new, height_new = (round(width * ratio), round(height * ratio))
img = img.resize((width_new, height_new))
img = img.convert('RGB')
img.save(image_filename, "PNG")
print(f"Resize image form {width}x{height} to {width_new}x{height_new}")
description = self.i2t.inference(image_filename)
Human_prompt = "\nHuman: provide a figure named {}. The description is: {}. This information helps you to understand this image, but you should use tools to finish following tasks, " \
"rather than directly imagine from my description. If you understand, say \"Received\". \n".format(image_filename, description)
AI_prompt = "Received. "
self.agent.memory.buffer = self.agent.memory.buffer + Human_prompt + 'AI: ' + AI_prompt
print("======>Current memory:\n %s" % self.agent.memory)
state = state + [(f"![](/file={image_filename})*{image_filename}*", AI_prompt)]
print("Outputs:", state)
return state, state, txt + ' ' + image_filename + ' ', temp_audio_filename
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if __name__ == '__main__':
bot = ConversationBot()
with gr.Blocks(css="#chatbot .overflow-y-auto{height:500px}") as demo:
with gr.Row():
gr.Markdown("## Audio ChatGPT")
chatbot = gr.Chatbot(elem_id="chatbot", label="Audio ChatGPT")
state = gr.State([])
with gr.Row():
with gr.Column(scale=0.7):
txt = gr.Textbox(show_label=False, placeholder="Enter text and press enter, or upload an image").style(container=False)
with gr.Column(scale=0.15, min_width=0):
clear = gr.Button("Clear")
with gr.Column(scale=0.15, min_width=0):
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btn = gr.UploadButton("Upload", file_types=["image","audio"])
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with gr.Column():
outaudio = gr.Audio()
txt.submit(bot.run_text, [txt, state], [chatbot, state, outaudio])
txt.submit(lambda: "", None, txt)
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#btn.upload(bot.run_image, [btn, state, txt], [chatbot, state, txt])
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btn.upload(bot.run_image_or_audio, [btn, state, txt], [chatbot, state, txt, outaudio])
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clear.click(bot.memory.clear)
clear.click(lambda: [], None, chatbot)
clear.click(lambda: [], None, state)
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#clear.click(lambda: [], None, outaudio)
demo.launch(server_name="0.0.0.0", server_port=7862, share=True)