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audio-chatgpt.py
1443 lines (1323 loc) · 77.1 KB
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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__))))
sys.path.append(os.path.join(os.path.dirname(os.path.realpath(__file__)), 'NeuralSeq'))
sys.path.append(os.path.join(os.path.dirname(os.path.realpath(__file__)), 'text_to_audio/Make_An_Audio'))
sys.path.append(os.path.join(os.path.dirname(os.path.realpath(__file__)), 'audio_detection'))
sys.path.append(os.path.join(os.path.dirname(os.path.realpath(__file__)), 'mono2binaural'))
import gradio as gr
import matplotlib
import librosa
import torch
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 PIL import Image
import numpy as np
from omegaconf import OmegaConf
from einops import repeat
from ldm.util import instantiate_from_config
from ldm.data.extract_mel_spectrogram import TRANSFORMS_16000
from vocoder.bigvgan.models import VocoderBigVGAN
from ldm.models.diffusion.ddim import DDIMSampler
import whisper
from utils.hparams import set_hparams
from utils.hparams import hparams as hp
import scipy.io.wavfile as wavfile
import librosa
from audio_infer.utils import config as detection_config
from audio_infer.pytorch.models import PVT
import clip
import numpy as np
AUDIO_CHATGPT_PREFIX = """AudioGPT
AudioGPT can not directly read audios, but it has a list of tools to finish different speech, audio, and singing voice tasks. Each audio will have a file name formed as "audio/xxx.wav". When talking about audios, AudioGPT is very strict to the file name and will never fabricate nonexistent files.
AudioGPT is able to use tools in a sequence, and is loyal to the tool observation outputs rather than faking the audio content and audio file name. It will remember to provide the file name from the last tool observation, if a new audio is generated.
Human may provide new audios to AudioGPT with a description. The description helps AudioGPT to understand this audio, but AudioGPT should use tools to finish following tasks, rather than directly imagine from the description.
Overall, AudioGPT is a powerful audio dialogue assistant tool that can help with a wide range of tasks and provide valuable insights and information on a wide range of topics.
TOOLS:
------
AudioGPT 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 audio 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}"""
def cut_dialogue_history(history_memory, keep_last_n_words = 500):
tokens = history_memory.split()
n_tokens = len(tokens)
print(f"history_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 merge_audio(audio_path_1, audio_path_2):
merged_signal = []
sr_1, signal_1 = wavfile.read(audio_path_1)
sr_2, signal_2 = wavfile.read(audio_path_2)
merged_signal.append(signal_1)
merged_signal.append(signal_2)
merged_signal = np.hstack(merged_signal)
merged_signal = np.asarray(merged_signal, dtype=np.int16)
audio_filename = os.path.join('audio', str(uuid.uuid4())[0:8] + ".wav")
wavfile.write(audio_filename, sr_2, merged_signal)
return audio_filename
class T2I:
def __init__(self, device):
from transformers import AutoModelForCausalLM, AutoTokenizer
from diffusers import StableDiffusionPipeline
from transformers import pipeline
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
class ImageCaptioning:
def __init__(self, device):
from transformers import BlipProcessor, BlipForConditionalGeneration
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
class T2A:
def __init__(self, device):
print("Initializing Make-An-Audio to %s" % device)
self.device = device
self.sampler = self._initialize_model('text_to_audio/Make_An_Audio/configs/text_to_audio/txt2audio_args.yaml', 'text_to_audio/Make_An_Audio/useful_ckpts/ta40multi_epoch=000085.ckpt', device=device)
self.vocoder = VocoderBigVGAN('text_to_audio/Make_An_Audio/vocoder/logs/bigv16k53w',device=device)
def _initialize_model(self, 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
def txt2audio(self, text, seed = 55, scale = 1.5, ddim_steps = 100, n_samples = 3, W = 624, H = 80):
SAMPLE_RATE = 16000
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 = self.select_best_audio(text, wav_list)
return best_wav
def select_best_audio(self, prompt, wav_list):
from wav_evaluation.models.CLAPWrapper import CLAPWrapper
clap_model = CLAPWrapper('text_to_audio/Make_An_Audio/useful_ckpts/CLAP/CLAP_weights_2022.pth', 'text_to_audio/Make_An_Audio/useful_ckpts/CLAP/config.yml',
use_cuda=torch.cuda.is_available())
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]
def inference(self, text, seed = 55, scale = 1.5, ddim_steps = 100, n_samples = 3, W = 624, H = 80):
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")
soundfile.write(audio_filename, result[1], samplerate = 16000)
print(f"Processed T2I.run, text: {text}, audio_filename: {audio_filename}")
return audio_filename
class I2A:
def __init__(self, device):
print("Initializing Make-An-Audio-Image to %s" % device)
self.device = device
self.sampler = self._initialize_model('text_to_audio/Make_An_Audio/configs/img_to_audio/img2audio_args.yaml', 'text_to_audio/Make_An_Audio/useful_ckpts/ta54_epoch=000216.ckpt', device=device)
self.vocoder = VocoderBigVGAN('text_to_audio/Make_An_Audio/vocoder/logs/bigv16k53w',device=device)
def _initialize_model(self, 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
def img2audio(self, image, seed = 55, scale = 3, ddim_steps = 100, W = 624, H = 80):
SAMPLE_RATE = 16000
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 = [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):
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")
soundfile.write(audio_filename, result[1], samplerate = 16000)
print(f"Processed I2a.run, image_filename: {image}, audio_filename: {audio_filename}")
return audio_filename
class TTS:
def __init__(self, device=None):
from inference.tts.PortaSpeech import TTSInference
if device is None:
device = 'cuda' if torch.cuda.is_available() else 'cpu'
print("Initializing PortaSpeech to %s" % device)
self.device = device
self.exp_name = 'checkpoints/ps_adv_baseline'
self.set_model_hparams()
self.inferencer = TTSInference(self.hp, device)
def set_model_hparams(self):
set_hparams(exp_name=self.exp_name, print_hparams=False)
self.hp = hp
def inference(self, text):
self.set_model_hparams()
inp = {"text": text}
out = self.inferencer.infer_once(inp)
audio_filename = os.path.join('audio', str(uuid.uuid4())[0:8] + ".wav")
soundfile.write(audio_filename, out, samplerate=22050)
return audio_filename
class T2S:
def __init__(self, device= None):
from inference.svs.ds_e2e import DiffSingerE2EInfer
if device is None:
device = 'cuda' if torch.cuda.is_available() else 'cpu'
print("Initializing DiffSinger to %s" % device)
self.device = device
self.exp_name = 'checkpoints/0831_opencpop_ds1000'
self.config= 'NeuralSeq/egs/egs_bases/svs/midi/e2e/opencpop/ds1000.yaml'
self.set_model_hparams()
self.pipe = DiffSingerE2EInfer(self.hp, device)
self.default_inp = {
'text': '你 说 你 不 SP 懂 为 何 在 这 时 牵 手 AP',
'notes': 'D#4/Eb4 | D#4/Eb4 | D#4/Eb4 | D#4/Eb4 | rest | D#4/Eb4 | D4 | D4 | D4 | D#4/Eb4 | F4 | D#4/Eb4 | D4 | rest',
'notes_duration': '0.113740 | 0.329060 | 0.287950 | 0.133480 | 0.150900 | 0.484730 | 0.242010 | 0.180820 | 0.343570 | 0.152050 | 0.266720 | 0.280310 | 0.633300 | 0.444590'
}
def set_model_hparams(self):
set_hparams(config=self.config, exp_name=self.exp_name, print_hparams=False)
self.hp = hp
def inference(self, inputs):
self.set_model_hparams()
val = inputs.split(",")
key = ['text', 'notes', 'notes_duration']
try:
inp = {k: v for k, v in zip(key, val)}
wav = self.pipe.infer_once(inp)
except:
print('Error occurs. Generate default audio sample.\n')
inp = self.default_inp
wav = self.pipe.infer_once(inp)
#if inputs == '' or len(val) < len(key):
# inp = self.default_inp
#else:
# inp = {k:v for k,v in zip(key,val)}
#wav = self.pipe.infer_once(inp)
wav *= 32767
audio_filename = os.path.join('audio', str(uuid.uuid4())[0:8] + ".wav")
wavfile.write(audio_filename, self.hp['audio_sample_rate'], wav.astype(np.int16))
print(f"Processed T2S.run, audio_filename: {audio_filename}")
return audio_filename
class t2s_VISinger:
def __init__(self, device=None):
from espnet2.bin.svs_inference import SingingGenerate
if device is None:
device = 'cuda' if torch.cuda.is_available() else 'cpu'
print("Initializing VISingere to %s" % device)
tag = 'AQuarterMile/opencpop_visinger1'
self.model = SingingGenerate.from_pretrained(
model_tag=str_or_none(tag),
device=device,
)
phn_dur = [[0. , 0.219 ],
[0.219 , 0.50599998],
[0.50599998, 0.71399999],
[0.71399999, 1.097 ],
[1.097 , 1.28799999],
[1.28799999, 1.98300004],
[1.98300004, 7.10500002],
[7.10500002, 7.60400009]]
phn = ['sh', 'i', 'q', 'v', 'n', 'i', 'SP', 'AP']
score = [[0, 0.50625, 'sh_i', 58, 'sh_i'], [0.50625, 1.09728, 'q_v', 56, 'q_v'], [1.09728, 1.9832100000000001, 'n_i', 53, 'n_i'], [1.9832100000000001, 7.105360000000001, 'SP', 0, 'SP'], [7.105360000000001, 7.604390000000001, 'AP', 0, 'AP']]
tempo = 70
tmp = {}
tmp["label"] = phn_dur, phn
tmp["score"] = tempo, score
self.default_inp = tmp
def inference(self, inputs):
val = inputs.split(",")
key = ['text', 'notes', 'notes_duration']
try: # TODO: input will be update
inp = {k: v for k, v in zip(key, val)}
wav = self.model(text=inp)["wav"]
except:
print('Error occurs. Generate default audio sample.\n')
inp = self.default_inp
wav = self.model(text=inp)["wav"]
audio_filename = os.path.join('audio', str(uuid.uuid4())[0:8] + ".wav")
soundfile.write(audio_filename, wav, samplerate=self.model.fs)
return audio_filename
class TTS_OOD:
def __init__(self, device):
from inference.tts.GenerSpeech import GenerSpeechInfer
if device is None:
device = 'cuda' if torch.cuda.is_available() else 'cpu'
print("Initializing GenerSpeech to %s" % device)
self.device = device
self.exp_name = 'checkpoints/GenerSpeech'
self.config = 'NeuralSeq/modules/GenerSpeech/config/generspeech.yaml'
self.set_model_hparams()
self.pipe = GenerSpeechInfer(self.hp, device)
def set_model_hparams(self):
set_hparams(config=self.config, exp_name=self.exp_name, print_hparams=False)
f0_stats_fn = f'{hp["binary_data_dir"]}/train_f0s_mean_std.npy'
if os.path.exists(f0_stats_fn):
hp['f0_mean'], hp['f0_std'] = np.load(f0_stats_fn)
hp['f0_mean'] = float(hp['f0_mean'])
hp['f0_std'] = float(hp['f0_std'])
hp['emotion_encoder_path'] = 'checkpoints/Emotion_encoder.pt'
self.hp = hp
def inference(self, inputs):
self.set_model_hparams()
key = ['ref_audio', 'text']
val = inputs.split(",")
inp = {k: v for k, v in zip(key, val)}
wav = self.pipe.infer_once(inp)
wav *= 32767
audio_filename = os.path.join('audio', str(uuid.uuid4())[0:8] + ".wav")
wavfile.write(audio_filename, self.hp['audio_sample_rate'], wav.astype(np.int16))
print(
f"Processed GenerSpeech.run. Input text:{val[1]}. Input reference audio: {val[0]}. Output Audio_filename: {audio_filename}")
return audio_filename
class Inpaint:
def __init__(self, device):
print("Initializing Make-An-Audio-inpaint to %s" % device)
self.device = device
self.sampler = self._initialize_model_inpaint('text_to_audio/Make_An_Audio/configs/inpaint/txt2audio_args.yaml', 'text_to_audio/Make_An_Audio/useful_ckpts/inpaint7_epoch00047.ckpt')
self.vocoder = VocoderBigVGAN('text_to_audio/Make_An_Audio/vocoder/logs/bigv16k53w',device=device)
self.cmap_transform = matplotlib.cm.viridis
def _initialize_model_inpaint(self, config, ckpt):
config = OmegaConf.load(config)
model = instantiate_from_config(config.model)
model.load_state_dict(torch.load(ckpt, map_location='cpu')["state_dict"], strict=False)
device = torch.device("cuda") if torch.cuda.is_available() else torch.device("cpu")
model = model.to(device)
print(model.device, device, model.cond_stage_model.device)
sampler = DDIMSampler(model)
return sampler
def make_batch_sd(self, 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(self, input_audio_path):
SAMPLE_RATE = 16000
sr, ori_wav = wavfile.read(input_audio_path)
print("gen_mel")
print(sr,ori_wav.shape,ori_wav)
ori_wav = ori_wav.astype(np.float32, order='C') / 32768.0
if len(ori_wav.shape)==2:# stereo
ori_wav = librosa.to_mono(ori_wav.T)
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 gen_mel_audio(self, input_audio):
SAMPLE_RATE = 16000
sr,ori_wav = input_audio
print("gen_mel_audio")
print(sr,ori_wav.shape,ori_wav)
ori_wav = ori_wav.astype(np.float32, order='C') / 32768.0
if len(ori_wav.shape)==2:# stereo
ori_wav = librosa.to_mono(ori_wav.T)
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(self, input_audio_path):
crop_len = 500
crop_mel = self.gen_mel(input_audio_path)[:,:crop_len]
color_mel = self.cmap_transform(crop_mel)
image = Image.fromarray((color_mel*255).astype(np.uint8))
image_filename = os.path.join('image', str(uuid.uuid4())[0:8] + ".png")
image.save(image_filename)
return image_filename
def inpaint(self, 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)
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 inference(self, input_audio, mel_and_mask, seed = 55, ddim_steps = 100):
SAMPLE_RATE = 16000
torch.set_grad_enabled(False)
mel_img = Image.open(mel_and_mask['image'])
mask_img = Image.open(mel_and_mask["mask"])
show_mel = np.array(mel_img.convert("L"))/255
mask = np.array(mask_img.convert("L"))/255
mel_bins,mel_len = 80,848
input_mel = self.gen_mel_audio(input_audio)[:,:mel_len]
mask = np.pad(mask,((0,0),(0,mel_len-mask.shape[1])),mode='constant',constant_values=0)
print(mask.shape,input_mel.shape)
with torch.no_grad():
batch = self.make_batch_sd(input_mel,mask,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 = self.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]
image = Image.fromarray((color_mel*255).astype(np.uint8))
image_filename = os.path.join('image', str(uuid.uuid4())[0:8] + ".png")
image.save(image_filename)
audio_filename = os.path.join('audio', str(uuid.uuid4())[0:8] + ".wav")
soundfile.write(audio_filename, gen_wav, samplerate = 16000)
return image_filename, audio_filename
class ASR:
def __init__(self, device):
print("Initializing Whisper to %s" % device)
self.device = device
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)
options = whisper.DecodingOptions()
result = whisper.decode(self.model, mel, options)
return result.text
def translate_english(self, audio_path):
audio = self.model.transcribe(audio_path, language='English')
return audio['text']
class A2T:
def __init__(self, device):
from audio_to_text.inference_waveform import AudioCapModel
print("Initializing Audio-To-Text Model to %s" % device)
self.device = device
self.model = AudioCapModel("audio_to_text/audiocaps_cntrstv_cnn14rnn_trm")
def inference(self, audio_path):
audio = whisper.load_audio(audio_path)
caption_text = self.model(audio)
return caption_text[0]
class GeneFace:
def __init__(self, device=None):
print("Initializing GeneFace model to %s" % device)
from audio_to_face.GeneFace_binding import GeneFaceInfer
if device is None:
device = 'cuda' if torch.cuda.is_available() else 'cpu'
self.device = device
self.geneface_model = GeneFaceInfer(device)
print("Loaded GeneFace model")
def inference(self, audio_path):
audio_base_name = os.path.basename(audio_path)[:-4]
out_video_name = audio_path.replace("audio","video").replace(".wav", ".mp4")
inp = {
'audio_source_name': audio_path,
'out_npy_name': f'geneface/tmp/{audio_base_name}.npy',
'cond_name': f'geneface/tmp/{audio_base_name}.npy',
'out_video_name': out_video_name,
'tmp_imgs_dir': f'video/tmp_imgs',
}
self.geneface_model.infer_once(inp)
return out_video_name
class SoundDetection:
def __init__(self, device):
self.device = device
self.sample_rate = 32000
self.window_size = 1024
self.hop_size = 320
self.mel_bins = 64
self.fmin = 50
self.fmax = 14000
self.model_type = 'PVT'
self.checkpoint_path = 'audio_detection/audio_infer/useful_ckpts/audio_detection.pth'
self.classes_num = detection_config.classes_num
self.labels = detection_config.labels
self.frames_per_second = self.sample_rate // self.hop_size
# Model = eval(self.model_type)
self.model = PVT(sample_rate=self.sample_rate, window_size=self.window_size,
hop_size=self.hop_size, mel_bins=self.mel_bins, fmin=self.fmin, fmax=self.fmax,
classes_num=self.classes_num)
checkpoint = torch.load(self.checkpoint_path, map_location=self.device)
self.model.load_state_dict(checkpoint['model'])
self.model.to(device)
def inference(self, audio_path):
# Forward
(waveform, _) = librosa.core.load(audio_path, sr=self.sample_rate, mono=True)
waveform = waveform[None, :] # (1, audio_length)
waveform = torch.from_numpy(waveform)
waveform = waveform.to(self.device)
# Forward
with torch.no_grad():
self.model.eval()
batch_output_dict = self.model(waveform, None)
framewise_output = batch_output_dict['framewise_output'].data.cpu().numpy()[0]
"""(time_steps, classes_num)"""
# print('Sound event detection result (time_steps x classes_num): {}'.format(
# framewise_output.shape))
import numpy as np
import matplotlib.pyplot as plt
sorted_indexes = np.argsort(np.max(framewise_output, axis=0))[::-1]
top_k = 10 # Show top results
top_result_mat = framewise_output[:, sorted_indexes[0 : top_k]]
"""(time_steps, top_k)"""
# Plot result
stft = librosa.core.stft(y=waveform[0].data.cpu().numpy(), n_fft=self.window_size,
hop_length=self.hop_size, window='hann', center=True)
frames_num = stft.shape[-1]
fig, axs = plt.subplots(2, 1, sharex=True, figsize=(10, 4))
axs[0].matshow(np.log(np.abs(stft)), origin='lower', aspect='auto', cmap='jet')
axs[0].set_ylabel('Frequency bins')
axs[0].set_title('Log spectrogram')
axs[1].matshow(top_result_mat.T, origin='upper', aspect='auto', cmap='jet', vmin=0, vmax=1)
axs[1].xaxis.set_ticks(np.arange(0, frames_num, self.frames_per_second))
axs[1].xaxis.set_ticklabels(np.arange(0, frames_num / self.frames_per_second))
axs[1].yaxis.set_ticks(np.arange(0, top_k))
axs[1].yaxis.set_ticklabels(np.array(self.labels)[sorted_indexes[0 : top_k]])
axs[1].yaxis.grid(color='k', linestyle='solid', linewidth=0.3, alpha=0.3)
axs[1].set_xlabel('Seconds')
axs[1].xaxis.set_ticks_position('bottom')
plt.tight_layout()
image_filename = os.path.join('image', str(uuid.uuid4())[0:8] + ".png")
plt.savefig(image_filename)
return image_filename
class SoundExtraction:
def __init__(self, device):
from sound_extraction.model.LASSNet import LASSNet
from sound_extraction.utils.stft import STFT
import torch.nn as nn
self.device = device
self.model_file = 'sound_extraction/useful_ckpts/LASSNet.pt'
self.stft = STFT()
self.model = nn.DataParallel(LASSNet(device)).to(device)
checkpoint = torch.load(self.model_file)
self.model.load_state_dict(checkpoint['model'])
self.model.eval()
def inference(self, inputs):
#key = ['ref_audio', 'text']
from sound_extraction.utils.wav_io import load_wav, save_wav
val = inputs.split(",")
audio_path = val[0] # audio_path, text
text = val[1]
waveform = load_wav(audio_path)
waveform = torch.tensor(waveform).transpose(1,0)
mixed_mag, mixed_phase = self.stft.transform(waveform)
text_query = ['[CLS] ' + text]
mixed_mag = mixed_mag.transpose(2,1).unsqueeze(0).to(self.device)
est_mask = self.model(mixed_mag, text_query)
est_mag = est_mask * mixed_mag
est_mag = est_mag.squeeze(1)
est_mag = est_mag.permute(0, 2, 1)
est_wav = self.stft.inverse(est_mag.cpu().detach(), mixed_phase)
est_wav = est_wav.squeeze(0).squeeze(0).numpy()
#est_path = f'output/est{i}.wav'
audio_filename = os.path.join('audio', str(uuid.uuid4())[0:8] + ".wav")
print('audio_filename ', audio_filename)
save_wav(est_wav, audio_filename)
return audio_filename
class Binaural:
def __init__(self, device):
from src.models import BinauralNetwork
self.device = device
self.model_file = 'mono2binaural/useful_ckpts/m2b/binaural_network.net'
self.position_file = ['mono2binaural/useful_ckpts/m2b/tx_positions.txt',
'mono2binaural/useful_ckpts/m2b/tx_positions2.txt',
'mono2binaural/useful_ckpts/m2b/tx_positions3.txt',
'mono2binaural/useful_ckpts/m2b/tx_positions4.txt',
'mono2binaural/useful_ckpts/m2b/tx_positions5.txt']
self.net = BinauralNetwork(view_dim=7,
warpnet_layers=4,
warpnet_channels=64,
)
self.net.load_from_file(self.model_file)
self.sr = 48000
def inference(self, audio_path):
mono, sr = librosa.load(path=audio_path, sr=self.sr, mono=True)
mono = torch.from_numpy(mono)
mono = mono.unsqueeze(0)
import numpy as np
import random
rand_int = random.randint(0,4)
view = np.loadtxt(self.position_file[rand_int]).transpose().astype(np.float32)
view = torch.from_numpy(view)
if not view.shape[-1] * 400 == mono.shape[-1]:
mono = mono[:,:(mono.shape[-1]//400)*400] #
if view.shape[1]*400 > mono.shape[1]:
m_a = view.shape[1] - mono.shape[-1]//400
rand_st = random.randint(0,m_a)
view = view[:,m_a:m_a+(mono.shape[-1]//400)] #
# binauralize and save output
self.net.eval().to(self.device)
mono, view = mono.to(self.device), view.to(self.device)
chunk_size = 48000 # forward in chunks of 1s
rec_field = 1000 # add 1000 samples as "safe bet" since warping has undefined rec. field
rec_field -= rec_field % 400 # make sure rec_field is a multiple of 400 to match audio and view frequencies
chunks = [
{
"mono": mono[:, max(0, i-rec_field):i+chunk_size],
"view": view[:, max(0, i-rec_field)//400:(i+chunk_size)//400]
}
for i in range(0, mono.shape[-1], chunk_size)
]
for i, chunk in enumerate(chunks):
with torch.no_grad():
mono = chunk["mono"].unsqueeze(0)
view = chunk["view"].unsqueeze(0)
binaural = self.net(mono, view).squeeze(0)
if i > 0:
binaural = binaural[:, -(mono.shape[-1]-rec_field):]
chunk["binaural"] = binaural
binaural = torch.cat([chunk["binaural"] for chunk in chunks], dim=-1)
binaural = torch.clamp(binaural, min=-1, max=1).cpu()
#binaural = chunked_forwarding(net, mono, view)
audio_filename = os.path.join('audio', str(uuid.uuid4())[0:8] + ".wav")
import torchaudio
torchaudio.save(audio_filename, binaural, sr)
#soundfile.write(audio_filename, binaural, samplerate = 48000)
print(f"Processed Binaural.run, audio_filename: {audio_filename}")
return audio_filename
class TargetSoundDetection:
def __init__(self, device):
from target_sound_detection.src import models as tsd_models
from target_sound_detection.src.models import event_labels
self.device = device
self.MEL_ARGS = {
'n_mels': 64,
'n_fft': 2048,
'hop_length': int(22050 * 20 / 1000),
'win_length': int(22050 * 40 / 1000)
}
self.EPS = np.spacing(1)
self.clip_model, _ = clip.load("ViT-B/32", device=self.device)
self.event_labels = event_labels
self.id_to_event = {i : label for i, label in enumerate(self.event_labels)}
config = torch.load('audio_detection/target_sound_detection/useful_ckpts/tsd/run_config.pth', map_location='cpu')
config_parameters = dict(config)
config_parameters['tao'] = 0.6
if 'thres' not in config_parameters.keys():
config_parameters['thres'] = 0.5
if 'time_resolution' not in config_parameters.keys():
config_parameters['time_resolution'] = 125
model_parameters = torch.load('audio_detection/target_sound_detection/useful_ckpts/tsd/run_model_7_loss=-0.0724.pt'
, map_location=lambda storage, loc: storage) # load parameter
self.model = getattr(tsd_models, config_parameters['model'])(config_parameters,
inputdim=64, outputdim=2, time_resolution=config_parameters['time_resolution'], **config_parameters['model_args'])
self.model.load_state_dict(model_parameters)
self.model = self.model.to(self.device).eval()
self.re_embeds = torch.load('audio_detection/target_sound_detection/useful_ckpts/tsd/text_emb.pth')
self.ref_mel = torch.load('audio_detection/target_sound_detection/useful_ckpts/tsd/ref_mel.pth')
def extract_feature(self, fname):
import soundfile as sf
y, sr = sf.read(fname, dtype='float32')
print('y ', y.shape)
ti = y.shape[0]/sr
if y.ndim > 1:
y = y.mean(1)
y = librosa.resample(y, sr, 22050)
lms_feature = np.log(librosa.feature.melspectrogram(y, **self.MEL_ARGS) + self.EPS).T
return lms_feature,ti
def build_clip(self, text):
text = clip.tokenize(text).to(self.device) # ["a diagram with dog", "a dog", "a cat"]
text_features = self.clip_model.encode_text(text)
return text_features
def cal_similarity(self, target, retrievals):
ans = []
#target =torch.from_numpy(target)
for name in retrievals.keys():
tmp = retrievals[name]
#tmp = torch.from_numpy(tmp)
s = torch.cosine_similarity(target.squeeze(), tmp.squeeze(), dim=0)
ans.append(s.item())
return ans.index(max(ans))
def inference(self, text, audio_path):
from target_sound_detection.src.utils import median_filter, decode_with_timestamps
target_emb = self.build_clip(text) # torch type
idx = self.cal_similarity(target_emb, self.re_embeds)
target_event = self.id_to_event[idx]
embedding = self.ref_mel[target_event]
embedding = torch.from_numpy(embedding)
embedding = embedding.unsqueeze(0).to(self.device).float()
#print('embedding ', embedding.shape)
inputs,ti = self.extract_feature(audio_path)
#print('ti ', ti)
inputs = torch.from_numpy(inputs)
inputs = inputs.unsqueeze(0).to(self.device).float()
#print('inputs ', inputs.shape)
decision, decision_up, logit = self.model(inputs, embedding)
pred = decision_up.detach().cpu().numpy()
pred = pred[:,:,0]
frame_num = decision_up.shape[1]
time_ratio = ti / frame_num
filtered_pred = median_filter(pred, window_size=1, threshold=0.5)
#print('filtered_pred ', filtered_pred)
time_predictions = []
for index_k in range(filtered_pred.shape[0]):
decoded_pred = []
decoded_pred_ = decode_with_timestamps(target_event, filtered_pred[index_k,:])
if len(decoded_pred_) == 0: # neg deal
decoded_pred_.append((target_event, 0, 0))
decoded_pred.append(decoded_pred_)
for num_batch in range(len(decoded_pred)): # when we test our model,the batch_size is 1
cur_pred = pred[num_batch]
# Save each frame output, for later visualization
label_prediction = decoded_pred[num_batch] # frame predict
# print(label_prediction)
for event_label, onset, offset in label_prediction:
time_predictions.append({
'onset': onset*time_ratio,
'offset': offset*time_ratio,})
ans = ''
for i,item in enumerate(time_predictions):
ans = ans + 'segment' + str(i+1) + ' start_time: ' + str(item['onset']) + ' end_time: ' + str(item['offset']) + '\t'
#print(ans)
return ans
# class Speech_Enh_SS_SC:
# """Speech Enhancement or Separation in single-channel
# Example usage:
# enh_model = Speech_Enh_SS("cuda")
# enh_wav = enh_model.inference("./test_chime4_audio_M05_440C0213_PED_REAL.wav")
# """
# def __init__(self, device="cuda", model_name="lichenda/chime4_fasnet_dprnn_tac"):
# self.model_name = model_name
# self.device = device
# print("Initializing ESPnet Enh to %s" % device)
# self._initialize_model()
# def _initialize_model(self):
# from espnet_model_zoo.downloader import ModelDownloader
# from espnet2.bin.enh_inference import SeparateSpeech
# d = ModelDownloader()
# cfg = d.download_and_unpack(self.model_name)
# self.separate_speech = SeparateSpeech(
# train_config=cfg["train_config"],
# model_file=cfg["model_file"],
# # for segment-wise process on long speech
# segment_size=2.4,
# hop_size=0.8,
# normalize_segment_scale=False,
# show_progressbar=True,
# ref_channel=None,
# normalize_output_wav=True,
# device=self.device,
# )
# def inference(self, speech_path, ref_channel=0):
# speech, sr = soundfile.read(speech_path)
# speech = speech[:, ref_channel]
# assert speech.dim() == 1
# enh_speech = self.separate_speech(speech[None, ], fs=sr)
# if len(enh_speech) == 1:
# return enh_speech[0]
# return enh_speech
# class Speech_Enh_SS_MC:
# """Speech Enhancement or Separation in multi-channel"""
# def __init__(self, device="cuda", model_name=None, ref_channel=4):
# self.model_name = model_name
# self.ref_channel = ref_channel
# self.device = device
# print("Initializing ESPnet Enh to %s" % device)
# self._initialize_model()
# def _initialize_model(self):
# from espnet_model_zoo.downloader import ModelDownloader
# from espnet2.bin.enh_inference import SeparateSpeech
# d = ModelDownloader()
# cfg = d.download_and_unpack(self.model_name)
# self.separate_speech = SeparateSpeech(
# train_config=cfg["train_config"],
# model_file=cfg["model_file"],
# # for segment-wise process on long speech
# segment_size=2.4,
# hop_size=0.8,
# normalize_segment_scale=False,
# show_progressbar=True,
# ref_channel=self.ref_channel,
# normalize_output_wav=True,
# device=self.device,
# )
# def inference(self, speech_path):
# speech, sr = soundfile.read(speech_path)
# speech = speech.T
# enh_speech = self.separate_speech(speech[None, ...], fs=sr)
# if len(enh_speech) == 1:
# return enh_speech[0]
# return enh_speech
class Speech_Enh_SS_SC:
"""Speech Enhancement or Separation in single-channel
Example usage:
enh_model = Speech_Enh_SS("cuda")
enh_wav = enh_model.inference("./test_chime4_audio_M05_440C0213_PED_REAL.wav")
"""
def __init__(self, device="cuda", model_name="espnet/Wangyou_Zhang_chime4_enh_train_enh_conv_tasnet_raw"):
self.model_name = model_name
self.device = device
print("Initializing ESPnet Enh to %s" % device)
self._initialize_model()
def _initialize_model(self):
from espnet_model_zoo.downloader import ModelDownloader
from espnet2.bin.enh_inference import SeparateSpeech
d = ModelDownloader()
cfg = d.download_and_unpack(self.model_name)
self.separate_speech = SeparateSpeech(
train_config=cfg["train_config"],
model_file=cfg["model_file"],
# for segment-wise process on long speech
segment_size=2.4,
hop_size=0.8,
normalize_segment_scale=False,
show_progressbar=True,
ref_channel=None,
normalize_output_wav=True,
device=self.device,
)
def inference(self, speech_path, ref_channel=0):
speech, sr = soundfile.read(speech_path)
speech = speech[:, ref_channel]
# speech = torch.from_numpy(speech)
# assert speech.dim() == 1
enh_speech = self.separate_speech(speech[None, ...], fs=sr)
audio_filename = os.path.join('audio', str(uuid.uuid4())[0:8] + ".wav")
# if len(enh_speech) == 1:
soundfile.write(audio_filename, enh_speech[0].squeeze(), samplerate=sr)
# return enh_speech[0]
# return enh_speech
# else: