mini-omni-2 / app.py
gpt-omni
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7ba9b1d
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10.3 kB
"""A simple web interactive chat demo based on gradio."""
import os
import time
import gradio as gr
import numpy as np
import spaces
import torch
import os
import lightning as L
import torch
import time
import spaces
from snac import SNAC
from litgpt import Tokenizer
from litgpt.utils import (
num_parameters,
)
from litgpt.generate.base import (
generate_AA,
generate_ASR,
generate_TA,
generate_TT,
generate_AT,
generate_TA_BATCH,
)
from typing import Any, Literal, Optional
import soundfile as sf
from litgpt.model import GPT, Config
from lightning.fabric.utilities.load import _lazy_load as lazy_load
from utils.snac_utils import layershift, reconscruct_snac, reconstruct_tensors, get_time_str
from utils.snac_utils import get_snac
import whisper
from tqdm import tqdm
from huggingface_hub import snapshot_download
from litgpt.generate.base import sample
device = "cuda" if torch.cuda.is_available() else "cpu"
ckpt_dir = "./checkpoint"
OUT_CHUNK = 4096
OUT_RATE = 24000
OUT_CHANNELS = 1
# TODO
text_vocabsize = 151936
text_specialtokens = 64
audio_vocabsize = 4096
audio_specialtokens = 64
padded_text_vocabsize = text_vocabsize + text_specialtokens
padded_audio_vocabsize = audio_vocabsize + audio_specialtokens
_eot = text_vocabsize
_pad_t = text_vocabsize + 1
_input_t = text_vocabsize + 2
_answer_t = text_vocabsize + 3
_asr = text_vocabsize + 4
_eoa = audio_vocabsize
_pad_a = audio_vocabsize + 1
_input_a = audio_vocabsize + 2
_answer_a = audio_vocabsize + 3
_split = audio_vocabsize + 4
def download_model(ckpt_dir):
repo_id = "gpt-omni/mini-omni"
snapshot_download(repo_id, local_dir=ckpt_dir, revision="main")
if not os.path.exists(ckpt_dir):
print(f"checkpoint directory {ckpt_dir} not found, downloading from huggingface")
download_model(ckpt_dir)
snacmodel = SNAC.from_pretrained("hubertsiuzdak/snac_24khz").eval().to(device)
whispermodel = whisper.load_model("small").to(device)
whispermodel.eval()
text_tokenizer = Tokenizer(ckpt_dir)
# fabric = L.Fabric(devices=1, strategy="auto")
config = Config.from_file(ckpt_dir + "/model_config.yaml")
config.post_adapter = False
model = GPT(config, device=device)
state_dict = lazy_load(ckpt_dir + "/lit_model.pth")
model.load_state_dict(state_dict, strict=True)
model = model.to(device)
model.eval()
@spaces.GPU
def get_input_ids_whisper_ATBatch(mel, leng, whispermodel, device):
# with torch.no_grad():
mel = mel.unsqueeze(0).to(device)
# audio_feature = whisper.decode(whispermodel,mel, options).audio_features
audio_feature = whispermodel.embed_audio(mel)[0][:leng]
T = audio_feature.size(0)
input_ids_AA = []
for i in range(7):
input_ids_item = []
input_ids_item.append(layershift(_input_a, i))
input_ids_item += [layershift(_pad_a, i)] * T
input_ids_item += [(layershift(_eoa, i)), layershift(_answer_a, i)]
input_ids_AA.append(torch.tensor(input_ids_item))
input_id_T = torch.tensor([_input_t] + [_pad_t] * T + [_eot, _answer_t])
input_ids_AA.append(input_id_T)
input_ids_AT = []
for i in range(7):
input_ids_item = []
input_ids_item.append(layershift(_input_a, i))
input_ids_item += [layershift(_pad_a, i)] * T
input_ids_item += [(layershift(_eoa, i)), layershift(_pad_a, i)]
input_ids_AT.append(torch.tensor(input_ids_item))
input_id_T = torch.tensor([_input_t] + [_pad_t] * T + [_eot, _answer_t])
input_ids_AT.append(input_id_T)
input_ids = [input_ids_AA, input_ids_AT]
stacked_inputids = [[] for _ in range(8)]
for i in range(2):
for j in range(8):
stacked_inputids[j].append(input_ids[i][j])
stacked_inputids = [torch.stack(tensors) for tensors in stacked_inputids]
return torch.stack([audio_feature, audio_feature]), stacked_inputids
@spaces.GPU
def next_token_batch(
model: GPT,
audio_features: torch.tensor,
input_ids: list,
whisper_lens: int,
task: list,
input_pos: torch.Tensor,
**kwargs: Any,
) -> torch.Tensor:
input_pos = input_pos.to(model.device)
input_ids = [input_id.to(model.device) for input_id in input_ids]
logits_a, logit_t = model(
audio_features, input_ids, input_pos, whisper_lens=whisper_lens, task=task
)
for i in range(7):
logits_a[i] = logits_a[i][0].unsqueeze(0)
logit_t = logit_t[1].unsqueeze(0)
next_audio_tokens = []
for logit_a in logits_a:
next_a = sample(logit_a, **kwargs).to(dtype=input_ids[0].dtype)
next_audio_tokens.append(next_a)
next_t = sample(logit_t, **kwargs).to(dtype=input_ids[0].dtype)
return next_audio_tokens, next_t
def load_audio(path):
audio = whisper.load_audio(path)
duration_ms = (len(audio) / 16000) * 1000
audio = whisper.pad_or_trim(audio)
mel = whisper.log_mel_spectrogram(audio)
return mel, int(duration_ms / 20) + 1
@spaces.GPU
def generate_audio_data(snac_tokens, snacmodel, device=None):
audio = reconstruct_tensors(snac_tokens, device)
with torch.inference_mode():
audio_hat = snacmodel.decode(audio)
audio_data = audio_hat.cpu().numpy().astype(np.float64) * 32768.0
audio_data = audio_data.astype(np.int16)
audio_data = audio_data.tobytes()
return audio_data
@torch.inference_mode()
def run_AT_batch_stream(
audio_path,
stream_stride=4,
max_returned_tokens=2048,
temperature=0.9,
top_k=1,
top_p=1.0,
eos_id_a=_eoa,
eos_id_t=_eot,
):
assert os.path.exists(audio_path), f"audio file {audio_path} not found"
model.set_kv_cache(batch_size=2, device=device)
mel, leng = load_audio(audio_path)
audio_feature, input_ids = get_input_ids_whisper_ATBatch(mel, leng, whispermodel, device)
T = input_ids[0].size(1)
# device = input_ids[0].device
assert max_returned_tokens > T, f"max_returned_tokens {max_returned_tokens} should be greater than audio length {T}"
if model.max_seq_length < max_returned_tokens - 1:
raise NotImplementedError(
f"max_seq_length {model.max_seq_length} needs to be >= {max_returned_tokens - 1}"
)
input_pos = torch.tensor([T], device=device)
list_output = [[] for i in range(8)]
tokens_A, token_T = next_token_batch(
model,
audio_feature.to(torch.float32).to(model.device),
input_ids,
[T - 3, T - 3],
["A1T2", "A1T2"],
input_pos=torch.arange(0, T, device=device),
temperature=temperature,
top_k=top_k,
top_p=top_p,
)
for i in range(7):
list_output[i].append(tokens_A[i].tolist()[0])
list_output[7].append(token_T.tolist()[0])
model_input_ids = [[] for i in range(8)]
for i in range(7):
tokens_A[i] = tokens_A[i].clone() + padded_text_vocabsize + i * padded_audio_vocabsize
model_input_ids[i].append(tokens_A[i].clone().to(device).to(torch.int32))
model_input_ids[i].append(torch.tensor([layershift(4097, i)], device=device))
model_input_ids[i] = torch.stack(model_input_ids[i])
model_input_ids[-1].append(token_T.clone().to(torch.int32))
model_input_ids[-1].append(token_T.clone().to(torch.int32))
model_input_ids[-1] = torch.stack(model_input_ids[-1])
text_end = False
index = 1
nums_generate = stream_stride
begin_generate = False
current_index = 0
for _ in tqdm(range(2, max_returned_tokens - T + 1)):
tokens_A, token_T = next_token_batch(
model,
None,
model_input_ids,
None,
None,
input_pos=input_pos,
temperature=temperature,
top_k=top_k,
top_p=top_p,
)
if text_end:
token_T = torch.tensor([_pad_t], device=device)
if tokens_A[-1] == eos_id_a:
break
if token_T == eos_id_t:
text_end = True
for i in range(7):
list_output[i].append(tokens_A[i].tolist()[0])
list_output[7].append(token_T.tolist()[0])
model_input_ids = [[] for i in range(8)]
for i in range(7):
tokens_A[i] = tokens_A[i].clone() +padded_text_vocabsize + i * padded_audio_vocabsize
model_input_ids[i].append(tokens_A[i].clone().to(device).to(torch.int32))
model_input_ids[i].append(
torch.tensor([layershift(4097, i)], device=device)
)
model_input_ids[i] = torch.stack(model_input_ids[i])
model_input_ids[-1].append(token_T.clone().to(torch.int32))
model_input_ids[-1].append(token_T.clone().to(torch.int32))
model_input_ids[-1] = torch.stack(model_input_ids[-1])
if index == 7:
begin_generate = True
if begin_generate:
current_index += 1
if current_index == nums_generate:
current_index = 0
snac = get_snac(list_output, index, nums_generate)
audio_stream = generate_audio_data(snac, snacmodel, device)
yield audio_stream
input_pos = input_pos.add_(1)
index += 1
text = text_tokenizer.decode(torch.tensor(list_output[-1]))
print(f"text output: {text}")
model.clear_kv_cache()
return list_output
for chunk in run_AT_batch_stream('./data/samples/output1.wav'):
pass
def process_audio(audio):
filepath = audio
print(f"filepath: {filepath}")
if filepath is None:
return
cnt = 0
tik = time.time()
for chunk in run_AT_batch_stream(filepath):
# Convert chunk to numpy array
if cnt == 0:
print(f"first chunk time cost: {time.time() - tik:.3f}")
cnt += 1
audio_data = np.frombuffer(chunk, dtype=np.int16)
audio_data = audio_data.reshape(-1, OUT_CHANNELS)
yield OUT_RATE, audio_data.astype(np.int16)
demo = gr.Interface(
process_audio,
inputs=gr.Audio(type="filepath", label="Microphone"),
outputs=[gr.Audio(label="Response", streaming=True, autoplay=True)],
title="Chat Mini-Omni Demo",
# live=True,
)
demo.queue()
demo.launch()