SongFormer / modeling_songformer.py
ASLP-lab's picture
fix bug
b0d4960
import pdb
from typing import Tuple
import torch
import torch.nn as nn
from transformers import PreTrainedModel
import argparse
import importlib
import json
import math
import multiprocessing as mp
import os
import time
from argparse import Namespace
from pathlib import Path
# monkey patch to fix issues in msaf
import scipy
import numpy as np
scipy.inf = np.inf
import librosa
import torch
from ema_pytorch import EMA
from loguru import logger
from muq import MuQ
from musicfm.model.musicfm_25hz import MusicFM25Hz
from omegaconf import OmegaConf
from tqdm import tqdm
import torch
import torch.nn as nn
from transformers import PreTrainedModel
from transformers.modeling_outputs import CausalLMOutputWithPast
from configuration_songformer import SongFormerConfig
from model_config import ModelConfig
from model import Model
from omegaconf import OmegaConf
# MUSICFM_HOME_PATH = os.path.join("ckpts", "MusicFM")
MUSICFM_HOME_PATH = "/home/node59_tmpdata3/cbhao/SongFormer_kaiyuan_test/github_test/SongFormer/src/SongFormer/ckpts/MusicFM"
BEFORE_DOWNSAMPLING_FRAME_RATES = 25
AFTER_DOWNSAMPLING_FRAME_RATES = 8.333
DATASET_LABEL = "SongForm-HX-8Class"
DATASET_IDS = [5]
TIME_DUR = 420
INPUT_SAMPLING_RATE = 24000
from dataset.label2id import DATASET_ID_ALLOWED_LABEL_IDS, DATASET_LABEL_TO_DATASET_ID
from postprocessing.functional import postprocess_functional_structure
def rule_post_processing(msa_list):
if len(msa_list) <= 2:
return msa_list
result = msa_list.copy()
while len(result) > 2:
first_duration = result[1][0] - result[0][0]
if first_duration < 1.0 and len(result) > 2:
result[0] = (result[0][0], result[1][1])
result = [result[0]] + result[2:]
else:
break
while len(result) > 2:
last_label_duration = result[-1][0] - result[-2][0]
if last_label_duration < 1.0:
result = result[:-2] + [result[-1]]
else:
break
while len(result) > 2:
if result[0][1] == result[1][1] and result[1][0] <= 10.0:
result = [(result[0][0], result[0][1])] + result[2:]
else:
break
while len(result) > 2:
last_duration = result[-1][0] - result[-2][0]
if result[-2][1] == result[-3][1] and last_duration <= 10.0:
result = result[:-2] + [result[-1]]
else:
break
return result
class SongFormerModel(PreTrainedModel):
config_class = SongFormerConfig
def __init__(self, config: SongFormerConfig):
super().__init__(config)
device = "cpu"
root_dir = os.environ["SONGFORMER_LOCAL_DIR"]
with open(os.path.join(root_dir, "muq_config2.json"), "r") as f:
muq_config_file = OmegaConf.load(f)
# self.muq = MuQ.from_pretrained("OpenMuQ/MuQ-large-msd-iter", device_map=None)
self.muq = MuQ(muq_config_file)
self.musicfm = MusicFM25Hz(
is_flash=False,
stat_path=os.path.join(root_dir, "msd_stats.json"),
# model_path=os.path.join(MUSICFM_HOME_PATH, "pretrained_msd.pt"),
)
self.songformer = Model(ModelConfig())
num_classes = config.num_classes
dataset_id2label_mask = {}
for key, allowed_ids in DATASET_ID_ALLOWED_LABEL_IDS.items():
dataset_id2label_mask[key] = np.ones(config.num_classes, dtype=bool)
dataset_id2label_mask[key][allowed_ids] = False
self.num_classes = num_classes
self.dataset_id2label_mask = dataset_id2label_mask
self.config = config
def forward(self, input):
with torch.no_grad():
INPUT_SAMPLING_RATE = 24000
device = next(self.parameters()).device
# 如果为tensor或者是numpy
if isinstance(input, (torch.Tensor, np.ndarray)):
audio = torch.tensor(input).to(device)
elif os.path.exists(input):
wav, sr = librosa.load(input, sr=INPUT_SAMPLING_RATE)
audio = torch.tensor(wav).to(device)
else:
raise ValueError("input should be a tensor/numpy or a valid file path")
win_size = self.config.win_size
hop_size = self.config.hop_size
num_classes = self.config.num_classes
total_len = (
(audio.shape[0] // INPUT_SAMPLING_RATE) // TIME_DUR
) * TIME_DUR + TIME_DUR
total_frames = math.ceil(total_len * AFTER_DOWNSAMPLING_FRAME_RATES)
logits = {
"function_logits": np.zeros([total_frames, num_classes]),
"boundary_logits": np.zeros([total_frames]),
}
logits_num = {
"function_logits": np.zeros([total_frames, num_classes]),
"boundary_logits": np.zeros([total_frames]),
}
lens = 0
i = 0
while True:
start_idx = i * INPUT_SAMPLING_RATE
end_idx = min((i + win_size) * INPUT_SAMPLING_RATE, audio.shape[-1])
if start_idx >= audio.shape[-1]:
break
if end_idx - start_idx <= 1024:
continue
audio_seg = audio[start_idx:end_idx]
# MuQ embedding
muq_output = self.muq(audio_seg.unsqueeze(0), output_hidden_states=True)
muq_embd_420s = muq_output["hidden_states"][10]
del muq_output
torch.cuda.empty_cache()
# MusicFM embedding
_, musicfm_hidden_states = self.musicfm.get_predictions(
audio_seg.unsqueeze(0)
)
musicfm_embd_420s = musicfm_hidden_states[10]
del musicfm_hidden_states
torch.cuda.empty_cache()
wraped_muq_embd_30s = []
wraped_musicfm_embd_30s = []
for idx_30s in range(i, i + hop_size, 30):
start_idx_30s = idx_30s * INPUT_SAMPLING_RATE
end_idx_30s = min(
(idx_30s + 30) * INPUT_SAMPLING_RATE,
audio.shape[-1],
(i + hop_size) * INPUT_SAMPLING_RATE,
)
if start_idx_30s >= audio.shape[-1]:
break
if end_idx_30s - start_idx_30s <= 1024:
continue
wraped_muq_embd_30s.append(
self.muq(
audio[start_idx_30s:end_idx_30s].unsqueeze(0),
output_hidden_states=True,
)["hidden_states"][10]
)
torch.cuda.empty_cache()
wraped_musicfm_embd_30s.append(
self.musicfm.get_predictions(
audio[start_idx_30s:end_idx_30s].unsqueeze(0)
)[1][10]
)
torch.cuda.empty_cache()
wraped_muq_embd_30s = torch.concatenate(wraped_muq_embd_30s, dim=1)
wraped_musicfm_embd_30s = torch.concatenate(
wraped_musicfm_embd_30s, dim=1
)
all_embds = [
wraped_musicfm_embd_30s,
wraped_muq_embd_30s,
musicfm_embd_420s,
muq_embd_420s,
]
if len(all_embds) > 1:
embd_lens = [x.shape[1] for x in all_embds]
max_embd_len = max(embd_lens)
min_embd_len = min(embd_lens)
if abs(max_embd_len - min_embd_len) > 4:
raise ValueError(
f"Embedding shapes differ too much: {max_embd_len} vs {min_embd_len}"
)
for idx in range(len(all_embds)):
all_embds[idx] = all_embds[idx][:, :min_embd_len, :]
embd = torch.concatenate(all_embds, axis=-1)
dataset_label = DATASET_LABEL
dataset_ids = torch.Tensor(DATASET_IDS).to(device, dtype=torch.long)
msa_info, chunk_logits = self.songformer.infer(
input_embeddings=embd,
dataset_ids=dataset_ids,
label_id_masks=torch.Tensor(
self.dataset_id2label_mask[
DATASET_LABEL_TO_DATASET_ID[dataset_label]
]
)
.to(device, dtype=bool)
.unsqueeze(0)
.unsqueeze(0),
with_logits=True,
)
start_frame = int(i * AFTER_DOWNSAMPLING_FRAME_RATES)
end_frame = start_frame + min(
math.ceil(hop_size * AFTER_DOWNSAMPLING_FRAME_RATES),
chunk_logits["boundary_logits"][0].shape[0],
)
logits["function_logits"][start_frame:end_frame, :] += (
chunk_logits["function_logits"][0].detach().cpu().numpy()
)
logits["boundary_logits"][start_frame:end_frame] = (
chunk_logits["boundary_logits"][0].detach().cpu().numpy()
)
logits_num["function_logits"][start_frame:end_frame, :] += 1
logits_num["boundary_logits"][start_frame:end_frame] += 1
lens += end_frame - start_frame
i += hop_size
logits["function_logits"] /= logits_num["function_logits"]
logits["boundary_logits"] /= logits_num["boundary_logits"]
logits["function_logits"] = torch.from_numpy(
logits["function_logits"][:lens]
).unsqueeze(0)
logits["boundary_logits"] = torch.from_numpy(
logits["boundary_logits"][:lens]
).unsqueeze(0)
msa_infer_output = postprocess_functional_structure(logits, self.config)
assert msa_infer_output[-1][-1] == "end"
if not self.config.no_rule_post_processing:
msa_infer_output = rule_post_processing(msa_infer_output)
msa_json = []
for idx in range(len(msa_infer_output) - 1):
msa_json.append(
{
"label": msa_infer_output[idx][1],
"start": msa_infer_output[idx][0],
"end": msa_infer_output[idx + 1][0],
}
)
return msa_json
@staticmethod
def _fix_state_dict_key_on_load(key: str) -> Tuple[str, bool]:
"""Replace legacy parameter names with their modern equivalents. E.g. beta -> bias, gamma -> weight."""
# ---- begin: ignore muq ----
if key.startswith("muq."):
return key, False
# ---- end ---
# Rename LayerNorm beta & gamma params for some early models ported from Tensorflow (e.g. Bert)
# This rename is logged.
if key.endswith("LayerNorm.beta"):
return key.replace("LayerNorm.beta", "LayerNorm.bias"), True
if key.endswith("LayerNorm.gamma"):
return key.replace("LayerNorm.gamma", "LayerNorm.weight"), True
# Rename weight norm parametrizations to match changes across torch versions.
# Impacts a number of speech/wav2vec models. e.g. Hubert, Wav2Vec2, and others.
# This rename is not logged.
if hasattr(nn.utils.parametrizations, "weight_norm"):
if key.endswith("weight_g"):
return key.replace(
"weight_g", "parametrizations.weight.original0"
), True
if key.endswith("weight_v"):
return key.replace(
"weight_v", "parametrizations.weight.original1"
), True
else:
if key.endswith("parametrizations.weight.original0"):
return key.replace(
"parametrizations.weight.original0", "weight_g"
), True
if key.endswith("parametrizations.weight.original1"):
return key.replace(
"parametrizations.weight.original1", "weight_v"
), True
return key, False