languagebind-source / languagebind /audio /processing_audio.py
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compat: vendor _expand_mask/clip_loss, fix tokenizer init, alias functional_tensor, guard set_audio_backend
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import cv2
import numpy as np
import torch
import torchaudio
from torchvision import transforms
from transformers import ProcessorMixin, BatchEncoding
from transformers.image_processing_utils import BatchFeature
from torch.nn import functional as F
def make_list_of_images(x):
if not isinstance(x, list):
return [x]
return x
# torchaudio.set_audio_backend was deprecated and later removed; it is a no-op
# on modern torchaudio, so only call it when present (older versions).
if hasattr(torchaudio, "set_audio_backend"):
torchaudio.set_audio_backend("soundfile")
def torchaudio_loader(path):
return torchaudio.load(path)
def int16_to_float32_torch(x):
return (x / 32767.0).type(torch.float32)
def float32_to_int16_torch(x):
x = torch.clamp(x, min=-1., max=1.)
return (x * 32767.).type(torch.int16)
DEFAULT_AUDIO_FRAME_SHIFT_MS = 10
class AudioTransform:
def __init__(self, args):
self.sample_rate = args.audio_sample_rate
self.num_mel_bins = args.num_mel_bins
self.target_length = args.target_length
self.audio_mean = args.audio_mean
self.audio_std = args.audio_std
self.mean = []
self.std = []
# mean=-4.2677393
# std=4.5689974
# self.norm = transforms.Normalize(mean=self.audio_mean, std=self.audio_std)
def __call__(self, audio_data_and_origin_sr):
audio_data, origin_sr = audio_data_and_origin_sr
if self.sample_rate != origin_sr:
# print(audio_data.shape, origin_sr)
audio_data = torchaudio.functional.resample(audio_data, orig_freq=origin_sr, new_freq=self.sample_rate)
waveform_melspec = self.waveform2melspec(audio_data)
return waveform_melspec
def waveform2melspec(self, audio_data):
mel = self.get_mel(audio_data)
if mel.shape[0] > self.target_length:
# split to three parts
chunk_frames = self.target_length
total_frames = mel.shape[0]
ranges = np.array_split(list(range(0, total_frames - chunk_frames + 1)), 3)
# print('total_frames-chunk_frames:', total_frames-chunk_frames,
# 'len(audio_data):', len(audio_data),
# 'chunk_frames:', chunk_frames,
# 'total_frames:', total_frames)
if len(ranges[1]) == 0: # if the audio is too short, we just use the first chunk
ranges[1] = [0]
if len(ranges[2]) == 0: # if the audio is too short, we just use the first chunk
ranges[2] = [0]
# randomly choose index for each part
idx_front = np.random.choice(ranges[0])
idx_middle = np.random.choice(ranges[1])
idx_back = np.random.choice(ranges[2])
# idx_front = ranges[0][0] # fixed
# idx_middle = ranges[1][0]
# idx_back = ranges[2][0]
# select mel
mel_chunk_front = mel[idx_front:idx_front + chunk_frames, :]
mel_chunk_middle = mel[idx_middle:idx_middle + chunk_frames, :]
mel_chunk_back = mel[idx_back:idx_back + chunk_frames, :]
# print(total_frames, idx_front, idx_front + chunk_frames, idx_middle, idx_middle + chunk_frames, idx_back, idx_back + chunk_frames)
# stack
mel_fusion = torch.stack([mel_chunk_front, mel_chunk_middle, mel_chunk_back], dim=0)
elif mel.shape[0] < self.target_length: # padding if too short
n_repeat = int(self.target_length / mel.shape[0]) + 1
# print(self.target_length, mel.shape[0], n_repeat)
mel = mel.repeat(n_repeat, 1)[:self.target_length, :]
mel_fusion = torch.stack([mel, mel, mel], dim=0)
else: # if equal
mel_fusion = torch.stack([mel, mel, mel], dim=0)
mel_fusion = mel_fusion.transpose(1, 2) # [3, target_length, mel_bins] -> [3, mel_bins, target_length]
# self.mean.append(mel_fusion.mean())
# self.std.append(mel_fusion.std())
mel_fusion = (mel_fusion - self.audio_mean) / (self.audio_std * 2)
return mel_fusion
def get_mel(self, audio_data):
# mel shape: (n_mels, T)
audio_data -= audio_data.mean()
mel = torchaudio.compliance.kaldi.fbank(
audio_data,
htk_compat=True,
sample_frequency=self.sample_rate,
use_energy=False,
window_type="hanning",
num_mel_bins=self.num_mel_bins,
dither=0.0,
frame_length=25,
frame_shift=DEFAULT_AUDIO_FRAME_SHIFT_MS,
)
return mel # (T, n_mels)
def get_audio_transform(config):
config = config.vision_config
return AudioTransform(config)
def load_and_transform_audio(
audio_path,
transform,
):
waveform_and_sr = torchaudio_loader(audio_path)
audio_outputs = transform(waveform_and_sr)
return audio_outputs
class LanguageBindAudioProcessor(ProcessorMixin):
attributes = []
tokenizer_class = ("LanguageBindAudioTokenizer")
def __init__(self, config, tokenizer=None, **kwargs):
super().__init__(**kwargs)
self.config = config
self.transform = get_audio_transform(config)
self.image_processor = load_and_transform_audio
self.tokenizer = tokenizer
def __call__(self, images=None, text=None, context_length=77, return_tensors=None, **kwargs):
if text is None and images is None:
raise ValueError("You have to specify either text or images. Both cannot be none.")
if text is not None:
encoding = self.tokenizer(text, max_length=context_length, padding='max_length',
truncation=True, return_tensors=return_tensors, **kwargs)
if images is not None:
images = make_list_of_images(images)
image_features = [self.image_processor(image, self.transform) for image in images]
image_features = torch.stack(image_features)
if text is not None and images is not None:
encoding["pixel_values"] = image_features
return encoding
elif text is not None:
return encoding
else:
return {"pixel_values": image_features}
def batch_decode(self, skip_special_tokens=True, *args, **kwargs):
"""
This method forwards all its arguments to CLIPTokenizerFast's [`~PreTrainedTokenizer.batch_decode`]. Please
refer to the docstring of this method for more information.
"""
return self.tokenizer.batch_decode(*args, skip_special_tokens=skip_special_tokens, **kwargs)
def decode(self, skip_special_tokens=True, *args, **kwargs):
"""
This method forwards all its arguments to CLIPTokenizerFast's [`~PreTrainedTokenizer.decode`]. Please refer to
the docstring of this method for more information.
"""
return self.tokenizer.decode(*args, skip_special_tokens=skip_special_tokens, **kwargs)