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Merge branch 'main' into hzq
# Conflicts: # assets/7ef0ec0b.wav # audio-chatgpt.py # download.sh
This commit is contained in:
143
.gitignore
vendored
Executable file
143
.gitignore
vendored
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@@ -0,0 +1,143 @@
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# Generated files
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|
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|
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|
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|
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# data
|
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# Cython-generated C++ source files
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.vscode/ftp-sync.json
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experimental/*
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|
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|
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multirun
|
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outputs
|
||||
49
README.md
49
README.md
@@ -1,13 +1,38 @@
|
||||
---
|
||||
title: Make An Audio
|
||||
emoji: 😻
|
||||
colorFrom: green
|
||||
colorTo: indigo
|
||||
sdk: gradio
|
||||
sdk_version: 3.17.0
|
||||
app_file: app.py
|
||||
pinned: false
|
||||
---
|
||||
|
||||
Check out the configuration reference at https://huggingface.co/docs/hub/spaces-config-reference
|
||||
# AudioGPT
|
||||
|
||||
**AudioGPT** connects ChatGPT and a series of Audio Foundation Models to enable **sending** and **receiving** speech, sing, and audio during chatting.
|
||||
|
||||
|
||||
## Capability
|
||||
Here we list the capability of AudioGPT at this time. More supported models and tasks are comming soon.
|
||||
|
||||
| Task | Foundation Model | Status |
|
||||
|:-------------------------:|:--------------------------------:|:------:|
|
||||
| ----------Speech--------- | / | / |
|
||||
| Text-to-Speech | [FastSpeech](), [SyntaSpeech]() | WIP |
|
||||
| Neural Vocoding | [BigVGAN](), [FastDiff]() | WIP |
|
||||
| Style Transfer | [GenerSpeech]() | WIP |
|
||||
| Speech Recognition | [whisper]() | Yes |
|
||||
| ----------Sing--------- | / | |
|
||||
| Text-to-Sing | [DiffSinger]() | Yes |
|
||||
| ----------Audio--------- | / | |
|
||||
| Text-to-Audio | [Make-An-Audio]() | Yes |
|
||||
| Audio Inpainting | [Make-An-Audio]() | WIP |
|
||||
| Image-to-Audio | [Make-An-Audio]() | Yes |
|
||||
|
||||
|
||||
|
||||
## Internal Version Updates
|
||||
|
||||
3.23 Support Text-to-Sing\
|
||||
3.21 Support Image-to-Sing\
|
||||
3.19 Support Speech Recognition\
|
||||
3.17 Support Text-to-Audio
|
||||
|
||||
## Acknowledgement
|
||||
We appreciate the open source of the following projects:
|
||||
|
||||
[Visual ChatGPT](https://github.com/microsoft/visual-chatgpt)  
|
||||
[Hugging Face](https://github.com/huggingface)  
|
||||
[LangChain](https://github.com/hwchase17/langchain)  
|
||||
[Stable Diffusion](https://github.com/CompVis/stable-diffusion)  
|
||||
|
||||
BIN
assets/7ef0ec0b.wav
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BIN
assets/7ef0ec0b.wav
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Binary file not shown.
23
assets/PROMPT.md
Normal file
23
assets/PROMPT.md
Normal file
@@ -0,0 +1,23 @@
|
||||
# Prompt Example
|
||||
## Text-To-Image
|
||||
Input Example : Generate an image of a horse<br />
|
||||
Output:<br />
|
||||
<br />
|
||||
## Text-To-Audio
|
||||
Input Example : Generate an audio of a piano playing<br />
|
||||
Output:<br />
|
||||
<br />
|
||||
## Text-To-Sing
|
||||
Input example : please generate a piece of singing voice. Text sequence is 小酒窝长睫毛AP是你最美的记号. Note sequence is C#4/Db4 | F#4/Gb4 | G#4/Ab4 | A#4/Bb4 F#4/Gb4 | F#4/Gb4 C#4/Db4 | C#4/Db4 | rest | C#4/Db4 | A#4/Bb4 | G#4/Ab4 | A#4/Bb4 | G#4/Ab4 | F4 | C#4/Db4. Note duration sequence is 0.407140 | 0.376190 | 0.242180 | 0.509550 0.183420 | 0.315400 0.235020 | 0.361660 | 0.223070 | 0.377270 | 0.340550 | 0.299620 | 0.344510 | 0.283770 | 0.323390 | 0.360340.<br />
|
||||
Output:<br />
|
||||
<br />
|
||||
## Image-To-Audio
|
||||
First upload your image(.png)<br />
|
||||
Input Example : Generate the audio of this image<br />
|
||||
Output:<br />
|
||||
<br />
|
||||
## ASR
|
||||
First uploag your audio(.wav)<br />
|
||||
Input Example : Generate the text of this audio<br />
|
||||
Output:<br />
|
||||
<br />
|
||||
BIN
assets/asr.png
Normal file
BIN
assets/asr.png
Normal file
Binary file not shown.
|
After Width: | Height: | Size: 65 KiB |
BIN
assets/i2a-1.png
Normal file
BIN
assets/i2a-1.png
Normal file
Binary file not shown.
|
After Width: | Height: | Size: 564 KiB |
BIN
assets/i2a-2.png
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BIN
assets/i2a-2.png
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Binary file not shown.
|
After Width: | Height: | Size: 553 KiB |
BIN
assets/t2a.png
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BIN
assets/t2a.png
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Binary file not shown.
|
After Width: | Height: | Size: 43 KiB |
BIN
assets/t2i.png
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BIN
assets/t2i.png
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Binary file not shown.
|
After Width: | Height: | Size: 668 KiB |
BIN
assets/t2s.png
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BIN
assets/t2s.png
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Binary file not shown.
|
After Width: | Height: | Size: 77 KiB |
345
audio-chatgpt.py
345
audio-chatgpt.py
@@ -3,7 +3,8 @@ 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__)), 'text_to_sing/DiffSinger'))
|
||||
sys.path.append(os.path.join(os.path.dirname(os.path.realpath(__file__)), 'text-to-audio/MakeAnAudio'))
|
||||
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__)), 'text_to_audio/Make_An_Audio_img'))
|
||||
import gradio as gr
|
||||
from transformers import AutoModelForCausalLM, AutoTokenizer, CLIPSegProcessor, CLIPSegForImageSegmentation
|
||||
import torch
|
||||
@@ -28,11 +29,15 @@ import einops
|
||||
from pytorch_lightning import seed_everything
|
||||
import random
|
||||
from ldm.util import instantiate_from_config
|
||||
from ldm.data.extract_mel_spectrogram import TRANSFORMS_16000
|
||||
from pathlib import Path
|
||||
from vocoder.hifigan.modules import VocoderHifigan
|
||||
from vocoder.bigvgan.models import VocoderBigVGAN
|
||||
from ldm.models.diffusion.ddim import DDIMSampler
|
||||
from wav_evaluation.models.CLAPWrapper import CLAPWrapper
|
||||
from inference.svs.ds_e2e import DiffSingerE2EInfer
|
||||
import whisper
|
||||
|
||||
import torch
|
||||
from inference.svs.ds_e2e import DiffSingerE2EInfer
|
||||
from inference.tts.GenerSpeech import GenerSpeechInfer
|
||||
@@ -67,7 +72,7 @@ Thought: Do I need to use a tool? No
|
||||
"""
|
||||
|
||||
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.
|
||||
You will remember to provide the audio file name loyally if it's provided in the last tool observation.
|
||||
|
||||
Begin!
|
||||
|
||||
@@ -76,8 +81,8 @@ Previous conversation history:
|
||||
New input: {input}
|
||||
Thought: Do I need to use a tool? {agent_scratchpad}"""
|
||||
|
||||
SAMPLE_RATE = 16000
|
||||
temp_audio_filename = "audio/c00d9240.wav"
|
||||
#temp_audio_filename = "audio/c00d9240.wav"
|
||||
|
||||
|
||||
def cut_dialogue_history(history_memory, keep_last_n_words = 500):
|
||||
tokens = history_memory.split()
|
||||
@@ -120,12 +125,11 @@ def initialize_model(config, ckpt, 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())
|
||||
|
||||
def select_best_audio(prompt,wav_list):
|
||||
clap_model = CLAPWrapper('useful_ckpts/CLAP/CLAP_weights_2022.pth','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:
|
||||
@@ -185,6 +189,18 @@ class T2I:
|
||||
print(f"Processed T2I.run, text: {text}, image_filename: {image_filename}")
|
||||
return image_filename
|
||||
|
||||
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
|
||||
|
||||
class T2A:
|
||||
def __init__(self, device):
|
||||
@@ -194,6 +210,7 @@ class T2A:
|
||||
self.vocoder = VocoderHifigan('vocoder/logs/hifi_0127',device=device)
|
||||
|
||||
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)
|
||||
@@ -220,7 +237,6 @@ class T2A:
|
||||
return best_wav
|
||||
|
||||
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(
|
||||
@@ -229,12 +245,59 @@ class T2A:
|
||||
W = mel_len
|
||||
)
|
||||
audio_filename = os.path.join('audio', str(uuid.uuid4())[0:8] + ".wav")
|
||||
temp_audio_filename = audio_filename
|
||||
wavfile.write(audio_filename, 16000, result[1])
|
||||
#soundfile.write(audio_filename, result[1], samplerate = 16000)
|
||||
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 = initialize_model('text_to_audio/Make_An_Audio_img/configs/img_to_audio/img2audio_args.yaml', 'text_to_audio/Make_An_Audio_img/useful_ckpts/ta54_epoch=000216.ckpt', device=device)
|
||||
self.vocoder = VocoderBigVGAN('text_to_audio/Make_An_Audio_img/vocoder/logs/bigv16k53w',device=device)
|
||||
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:[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):
|
||||
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 T2S:
|
||||
def __init__(self, device= None):
|
||||
if device is None:
|
||||
@@ -256,7 +319,6 @@ class T2S:
|
||||
self.hp = hp
|
||||
|
||||
def inference(self, inputs):
|
||||
global temp_audio_filename
|
||||
self.set_model_hparams()
|
||||
val = inputs.split(",")
|
||||
key = ['text', 'notes', 'notes_duration']
|
||||
@@ -267,10 +329,9 @@ class T2S:
|
||||
wav = self.pipe.infer_once(inp)
|
||||
wav *= 32767
|
||||
audio_filename = os.path.join('audio', str(uuid.uuid4())[0:8] + ".wav")
|
||||
temp_audio_filename = audio_filename
|
||||
wavfile.write(temp_audio_filename, self.hp['audio_sample_rate'], wav.astype(np.int16))
|
||||
wavfile.write(audio_filename, self.hp['audio_sample_rate'], wav.astype(np.int16))
|
||||
print(f"Processed T2S.run, audio_filename: {audio_filename}")
|
||||
return temp_audio_filename
|
||||
return audio_filename
|
||||
|
||||
class TTS_OOD:
|
||||
def __init__(self, device):
|
||||
@@ -294,7 +355,6 @@ class TTS_OOD:
|
||||
self.hp = hp
|
||||
|
||||
def inference(self, inputs):
|
||||
global temp_audio_filename
|
||||
self.set_model_hparams()
|
||||
key = ['ref_audio', 'text']
|
||||
val = inputs.split(",")
|
||||
@@ -302,20 +362,145 @@ class TTS_OOD:
|
||||
wav = self.pipe.infer_once(inp)
|
||||
wav *= 32767
|
||||
audio_filename = os.path.join('audio', str(uuid.uuid4())[0:8] + ".wav")
|
||||
temp_audio_filename = audio_filename
|
||||
wavfile.write(temp_audio_filename, self.hp['audio_sample_rate'], wav.astype(np.int16))
|
||||
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 temp_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 = initialize_model('text_to_audio/Make_An_Audio_inpaint/configs/inpaint/txt2audio_args.yaml',
|
||||
'text_to_audio/Make_An_Audio_inpaint/useful_ckpts/inpaint7_epoch00047.ckpt')
|
||||
self.vocoder = VocoderBigVGAN('./vocoder/logs/bigv16k53w', device=device)
|
||||
|
||||
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):
|
||||
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(self, 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(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)
|
||||
|
||||
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(self, 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)
|
||||
|
||||
|
||||
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
|
||||
|
||||
class ConversationBot:
|
||||
def __init__(self):
|
||||
print("Initializing AudioChatGPT")
|
||||
self.llm = OpenAI(temperature=0)
|
||||
|
||||
self.t2i = T2I(device="cuda:0")
|
||||
self.i2t = ImageCaptioning(device="cuda:1")
|
||||
self.t2a = T2A(device="cuda:0")
|
||||
self.t2s = T2S(device="cuda:2")
|
||||
self.i2a = I2A(device="cuda:1")
|
||||
self.asr = ASR(device="cuda:1")
|
||||
self.t2s = T2S(device="cuda:0")
|
||||
self.tts_ood = TTS_OOD(device="cuda:0")
|
||||
self.memory = ConversationBufferMemory(memory_key="chat_history", output_key='output')
|
||||
@@ -323,6 +508,9 @@ class ConversationBot:
|
||||
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. "),
|
||||
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. "),
|
||||
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."
|
||||
"The input to this tool should be a string, representing the text used to generate audio."),
|
||||
@@ -337,7 +525,16 @@ class ConversationBot:
|
||||
"If Like: Generate a piece of singing voice, the input to this tool should be \"\" since there is no User Input Text, Note and Duration Sequence ."
|
||||
"If Like: Generate a piece of singing voice. Text: xxx, Note: xxx, Duration: xxx. "
|
||||
"Or Like: Generate a piece of singing voice. Text is xxx, note is xxx, duration is xxx."
|
||||
"The input to this tool should be a comma seperated string of three, representing text, note and duration sequence since User Input Text, Note and Duration Sequence are all provided.")]
|
||||
"The input to this tool should be a comma seperated string of three, representing text, note and duration sequence since User Input Text, Note and Duration Sequence are all provided."),
|
||||
Tool(name="Generate singing voice From User Input Text", func=self.t2s.inference,
|
||||
description="useful for when you want to generate a piece of singing voice from its description."
|
||||
"The input to this tool should be a comma seperated string of three, representing the text sequence and its corresponding note and duration sequence."),
|
||||
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. "),
|
||||
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.")]
|
||||
self.agent = initialize_agent(
|
||||
self.tools,
|
||||
self.llm,
|
||||
@@ -347,65 +544,69 @@ class ConversationBot:
|
||||
return_intermediate_steps=True,
|
||||
agent_kwargs={'prefix': AUDIO_CHATGPT_PREFIX, 'format_instructions': AUDIO_CHATGPT_FORMAT_INSTRUCTIONS, 'suffix': AUDIO_CHATGPT_SUFFIX}, )
|
||||
|
||||
def run_file(self, file, state, txt):
|
||||
if file.name.endswith('.wav') or file.name.endswith('.wav'):
|
||||
return self.run_audio(file, state, txt)
|
||||
else:
|
||||
return self.run_image(file, state, txt)
|
||||
|
||||
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})
|
||||
tool = res['intermediate_steps'][0][0].tool
|
||||
if tool == "Generate Image From User Input Text":
|
||||
print("======>Current memory:\n %s" % self.agent.memory)
|
||||
response = re.sub('(image/\S*png)', lambda m: f'})*{m.group(0)}*', res['output'])
|
||||
state = state + [(text, response)]
|
||||
print("Outputs:", state)
|
||||
return state, state, None
|
||||
print("======>Current memory:\n %s" % self.agent.memory)
|
||||
response = re.sub('(audio/\S*wav)', lambda m: f'})*{m.group(0)}*', res['output'])
|
||||
audio_filename = res['intermediate_steps'][0][1]
|
||||
response = re.sub('(image/\S*png)', lambda m: f'})*{m.group(0)}*', res['output'])
|
||||
#response = res['output'] + f"<audio src=audio_filename controls=controls></audio>"
|
||||
state = state + [(text, response)]
|
||||
print("Outputs:", state)
|
||||
return state, state
|
||||
return state, state, audio_filename
|
||||
|
||||
def run_image(self, image, state, txt):
|
||||
print("===============Running run_image =============")
|
||||
print("Inputs:", image, 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(image.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"*{image_filename}*", AI_prompt)]
|
||||
print("Outputs:", state)
|
||||
return state, state, f'{txt} {image_filename} '
|
||||
|
||||
def run_audio(self, audio, state, txt):
|
||||
print("===============Running run_audio =============")
|
||||
print("Inputs:", audio, state)
|
||||
print("======>Previous memory:\n %s" % self.agent.memory)
|
||||
audio_filename = os.path.join('audio', str(uuid.uuid4())[0:8] + ".wav")
|
||||
move_file(audio.name, audio_filename)
|
||||
Human_prompt = "\nHuman: provide an reference audio named {}. You use tools to finish following tasks, " \
|
||||
"rather than directly imagine from my description. If you understand, say \"Received\". \n".format(
|
||||
audio_filename)
|
||||
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"*{audio_filename}*", AI_prompt)]
|
||||
print("Outputs:", state)
|
||||
return state, state, f'{txt} {audio_filename} '
|
||||
|
||||
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)
|
||||
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 src=audio_filename controls=controls></audio>*{audio_filename}*", AI_prompt)]
|
||||
state = state + [(f"*{audio_filename}*", AI_prompt)]
|
||||
print("Outputs:", state)
|
||||
return state, state, txt + ' ' + audio_filename + ' ', 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"*{image_filename}*", AI_prompt)]
|
||||
print("Outputs:", state)
|
||||
return state, state, txt + ' ' + image_filename + ' ', None
|
||||
|
||||
if __name__ == '__main__':
|
||||
bot = ConversationBot()
|
||||
@@ -416,18 +617,16 @@ if __name__ == '__main__':
|
||||
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 audio").style(
|
||||
container=False)
|
||||
txt = gr.Textbox(show_label=False, placeholder="Enter text and press enter, or upload an image or audio").style(container=False)
|
||||
with gr.Column(scale=0.15, min_width=0):
|
||||
clear = gr.Button("Clear️")
|
||||
with gr.Column(scale=0.15, min_width=0):
|
||||
btn = gr.UploadButton("Upload", file_types=["audio", "image"])
|
||||
|
||||
btn = gr.UploadButton("Upload", file_types=["image","audio"])
|
||||
with gr.Column():
|
||||
outaudio = gr.Audio()
|
||||
txt.submit(bot.run_text, [txt, state], [chatbot, state, outaudio])
|
||||
txt.submit(lambda: "", None, txt)
|
||||
btn.upload(bot.run_file, [btn, state, txt], [chatbot, state, txt])
|
||||
btn.upload(bot.run_image_or_audio, [btn, state, txt], [chatbot, state, txt, outaudio])
|
||||
clear.click(bot.memory.clear)
|
||||
clear.click(lambda: [], None, chatbot)
|
||||
clear.click(lambda: [], None, state)
|
||||
|
||||
13
download.sh
13
download.sh
@@ -1,7 +1,20 @@
|
||||
mkdir checkpoints
|
||||
mkdir audio
|
||||
mkdir image
|
||||
mkdir text_to_audio
|
||||
wget -P checkpoints/0831_opencpop_ds1000/ -i https://huggingface.co/spaces/Silentlin/DiffSinger/resolve/main/checkpoints/0831_opencpop_ds1000/config.yaml https://huggingface.co/spaces/Silentlin/DiffSinger/resolve/main/checkpoints/0831_opencpop_ds1000/model_ckpt_steps_320000.ckpt
|
||||
wget -P checkpoints/0109_hifigan_bigpopcs_hop128/ -i https://huggingface.co/spaces/Silentlin/DiffSinger/blob/main/checkpoints/0109_hifigan_bigpopcs_hop128/config.yaml https://huggingface.co/spaces/Silentlin/DiffSinger/resolve/main/checkpoints/0109_hifigan_bigpopcs_hop128/model_ckpt_steps_1512000.ckpt
|
||||
wget -P checkpoints/0102_xiaoma_pe/ -i https://huggingface.co/spaces/Silentlin/DiffSinger/blob/main/checkpoints/0102_xiaoma_pe/config.yaml https://huggingface.co/spaces/Silentlin/DiffSinger/resolve/main/checkpoints/0102_xiaoma_pe/model_ckpt_steps_60000.ckpt
|
||||
cd text_to_audio
|
||||
git clone https://huggingface.co/spaces/DiffusionSpeech/Make_An_Audio
|
||||
git clone https://huggingface.co/spaces/DiffusionSpeech/Make_An_Audio_img
|
||||
git clone https://huggingface.co/spaces/DiffusionSpeech/Make_An_Audio_inpaint
|
||||
wget -P text_to_audio/Make_An_Audio/useful_ckpts/ -i https://huggingface.co/spaces/DiffusionSpeech/Make_An_Audio/resolve/main/useful_ckpts/ta40multi_epoch=000085.ckpt
|
||||
wget -P text_to_audio/Make_An_Audio/useful_ckpts/CLAP/ -i https://huggingface.co/spaces/DiffusionSpeech/Make_An_Audio/resolve/main/useful_ckpts/CLAP/CLAP_weights_2022.pth
|
||||
wget -P text_to_audio/Make_An_Audio_img/useful_ckpts/ -i https://huggingface.co/spaces/DiffusionSpeech/Make_An_Audio_img/resolve/main/useful_ckpts/ta54_epoch=000216.ckpt
|
||||
wget -P text_to_audio/Make_An_Audio_img/useful_ckpts/CLAP/ -i https://huggingface.co/spaces/DiffusionSpeech/Make_An_Audio_img/blob/main/useful_ckpts/CLAP/CLAP_weights_2022.pth
|
||||
wget -P text_to_audio/Make_An_Audio_inpaint/useful_ckpts/ -i https://huggingface.co/spaces/DiffusionSpeech/Make_An_Audio_inpaint/resolve/main/useful_ckpts/inpaint7_epoch00047.ckpt
|
||||
|
||||
wget -P checkpoints/GenerSpeech/ -i https://huggingface.co/spaces/Rongjiehuang/GenerSpeech/blob/main/checkpoints/GenerSpeech/config.yaml https://huggingface.co/spaces/Rongjiehuang/GenerSpeech/resolve/main/checkpoints/GenerSpeech/model_ckpt_steps_300000.ckpt
|
||||
wget -P checkpoints/trainset_hifigan/ -i https://huggingface.co/spaces/Rongjiehuang/GenerSpeech/blob/main/checkpoints/trainset_hifigan/config.yaml https://huggingface.co/spaces/Rongjiehuang/GenerSpeech/resolve/main/checkpoints/trainset_hifigan/model_ckpt_steps_1000000.ckpt
|
||||
wget -P checkpoints/ https://huggingface.co/spaces/Rongjiehuang/GenerSpeech/resolve/main/checkpoints/Emotion_encoder.pt
|
||||
|
||||
61
requirements.txt
Normal file
61
requirements.txt
Normal file
@@ -0,0 +1,61 @@
|
||||
--extra-index-url https://download.pytorch.org/whl/cu113
|
||||
accelerate
|
||||
addict==2.4.0
|
||||
albumentations==1.3.0
|
||||
appdirs==1.4.4
|
||||
basicsr==1.4.2
|
||||
beautifulsoup4==4.10.0
|
||||
Cython==0.29.24
|
||||
diffusers
|
||||
einops==0.3.0
|
||||
g2p-en==2.1.0
|
||||
google==3.0.0
|
||||
gradio
|
||||
h5py==2.8.0
|
||||
imageio==2.9.0
|
||||
imageio-ffmpeg==0.4.2
|
||||
invisible-watermark>=0.1.5
|
||||
kornia==0.6
|
||||
langchain==0.0.101
|
||||
librosa
|
||||
miditoolkit==0.1.7
|
||||
moviepy==1.0.3
|
||||
numpy==1.23.1
|
||||
omegaconf==2.1.1
|
||||
open_clip_torch==2.0.2
|
||||
openai
|
||||
openai-whisper
|
||||
opencv-contrib-python==4.3.0.36
|
||||
praat-parselmouth==0.3.3
|
||||
prettytable==3.6.0
|
||||
proglog==0.1.9
|
||||
pycwt==0.3.0a22
|
||||
pyloudnorm==0.1.0
|
||||
pypinyin==0.43.0
|
||||
pytorch-lightning==1.5.0
|
||||
pytorch-ssim==0.1
|
||||
pyworld==0.3.0
|
||||
resampy==0.2.2
|
||||
Resemblyzer==0.1.1.dev0
|
||||
safetensors==0.2.7
|
||||
sklearn==0.0
|
||||
soundfile
|
||||
soupsieve==2.3
|
||||
streamlit==1.12.1
|
||||
streamlit-drawable-canvas==0.8.0
|
||||
tensorboardX==2.4
|
||||
test-tube>=0.7.5
|
||||
TextGrid==1.5
|
||||
timm==0.6.12
|
||||
torch==1.12.1
|
||||
torchaudio==0.12.1
|
||||
torch-fidelity==0.3.0
|
||||
torchlibrosa
|
||||
torchmetrics==0.6.0
|
||||
torchvision==0.13.1
|
||||
transformers==4.26.1
|
||||
typing-extensions==3.10.0.2
|
||||
uuid==1.30
|
||||
webdataset==0.2.5
|
||||
webrtcvad==2.0.10
|
||||
yapf==0.32.0
|
||||
19
run.md
Normal file
19
run.md
Normal file
@@ -0,0 +1,19 @@
|
||||
# Run AudioGPT
|
||||
```
|
||||
# create a new environment
|
||||
conda create -n audiogpt python=3.8
|
||||
|
||||
# prepare the basic environments
|
||||
pip install -r requirements.txt
|
||||
|
||||
# download the foundation models you need
|
||||
bash download.sh
|
||||
|
||||
# prepare your private openAI private key
|
||||
export OPENAI_API_KEY={Your_Private_Openai_Key}
|
||||
|
||||
# Start AudioGPT !
|
||||
python audio-chatgpt.py
|
||||
```
|
||||
|
||||
|
||||
Reference in New Issue
Block a user