You can not select more than 25 topics
Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
323 lines
13 KiB
323 lines
13 KiB
|
1 week ago
|
# Copyright 2025 Sesame and The HuggingFace Inc. team. All rights reserved.
|
||
|
|
#
|
||
|
|
# Licensed under the Apache License, Version 2.0 (the "License");
|
||
|
|
# you may not use this file except in compliance with the License.
|
||
|
|
# You may obtain a copy of the License at
|
||
|
|
#
|
||
|
|
# http://www.apache.org/licenses/LICENSE-2.0
|
||
|
|
#
|
||
|
|
# Unless required by applicable law or agreed to in writing, software
|
||
|
|
# distributed under the License is distributed on an "AS IS" BASIS,
|
||
|
|
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
|
||
|
|
# See the License for the specific language governing permissions and
|
||
|
|
# limitations under the License.
|
||
|
|
|
||
|
|
import math
|
||
|
|
from pathlib import Path
|
||
|
|
from typing import Any
|
||
|
|
|
||
|
|
import numpy as np
|
||
|
|
|
||
|
|
from ...utils import auto_docstring, is_soundfile_available, is_torch_available
|
||
|
|
|
||
|
|
|
||
|
|
if is_torch_available():
|
||
|
|
import torch
|
||
|
|
|
||
|
|
if is_soundfile_available():
|
||
|
|
import soundfile as sf
|
||
|
|
|
||
|
|
from ...audio_utils import AudioInput, make_list_of_audio
|
||
|
|
from ...feature_extraction_utils import BatchFeature
|
||
|
|
from ...processing_utils import AudioKwargs, ProcessingKwargs, ProcessorMixin, Unpack
|
||
|
|
from ...tokenization_utils_base import PreTokenizedInput, TextInput
|
||
|
|
|
||
|
|
|
||
|
|
class CsmAudioKwargs(AudioKwargs, total=False):
|
||
|
|
"""
|
||
|
|
encoded_length_kwargs (`dict[str, Any]`, *optional*):
|
||
|
|
Dictionary of keyword arguments used to compute the encoded audio sequence length. This includes parameters
|
||
|
|
such as `kernel_sizes`, `strides`, `dilations`, and `use_causal_conv` that define the convolutional layers
|
||
|
|
used in audio encoding. The encoded length is used to determine how many audio tokens to generate for each
|
||
|
|
audio input in the text sequence.
|
||
|
|
"""
|
||
|
|
|
||
|
|
encoded_length_kwargs: dict[str, Any] | None
|
||
|
|
|
||
|
|
|
||
|
|
class CsmProcessorKwargs(ProcessingKwargs, total=False):
|
||
|
|
audio_kwargs: CsmAudioKwargs
|
||
|
|
_defaults = {
|
||
|
|
"text_kwargs": {
|
||
|
|
"padding": True,
|
||
|
|
"padding_side": "left",
|
||
|
|
"add_special_tokens": False,
|
||
|
|
},
|
||
|
|
"audio_kwargs": {
|
||
|
|
"encoded_length_kwargs": {
|
||
|
|
"kernel_sizes": [7, 3, 1, 8, 3, 1, 10, 3, 1, 12, 3, 1, 16, 3, 4],
|
||
|
|
"strides": [1, 1, 1, 4, 1, 1, 5, 1, 1, 6, 1, 1, 8, 1, 2],
|
||
|
|
"dilations": [1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1],
|
||
|
|
"use_causal_conv": True,
|
||
|
|
},
|
||
|
|
"sampling_rate": 24000,
|
||
|
|
},
|
||
|
|
"common_kwargs": {"return_tensors": "pt"},
|
||
|
|
}
|
||
|
|
|
||
|
|
|
||
|
|
@auto_docstring
|
||
|
|
class CsmProcessor(ProcessorMixin):
|
||
|
|
def __init__(
|
||
|
|
self,
|
||
|
|
feature_extractor,
|
||
|
|
tokenizer,
|
||
|
|
chat_template=None,
|
||
|
|
):
|
||
|
|
if not hasattr(tokenizer, "audio_token"):
|
||
|
|
self.audio_token = "<|AUDIO|>"
|
||
|
|
self.audio_token_id = tokenizer.convert_tokens_to_ids(self.audio_token)
|
||
|
|
else:
|
||
|
|
self.audio_token = tokenizer.audio_token
|
||
|
|
self.audio_token_id = tokenizer.audio_token_id
|
||
|
|
|
||
|
|
if not hasattr(tokenizer, "audio_eos_token"):
|
||
|
|
self.audio_eos_token = "<|audio_eos|>"
|
||
|
|
self.audio_eos_token_id = tokenizer.convert_tokens_to_ids(self.audio_eos_token)
|
||
|
|
else:
|
||
|
|
self.audio_eos_token = tokenizer.audio_eos_token
|
||
|
|
self.audio_eos_token_id = tokenizer.audio_eos_token_id
|
||
|
|
|
||
|
|
super().__init__(feature_extractor, tokenizer, chat_template=chat_template)
|
||
|
|
|
||
|
|
@staticmethod
|
||
|
|
def _get_encoded_length(audio_length, kernel_sizes=None, strides=None, dilations=None, use_causal_conv=None):
|
||
|
|
"""
|
||
|
|
Compute the length of the encoded audio sequence.
|
||
|
|
|
||
|
|
Args:
|
||
|
|
audio_length (int): The length of the audio sequence.
|
||
|
|
kernel_sizes (list[int]): The kernel sizes for the convolutional layers.
|
||
|
|
strides (list[int]): The strides for the convolutional layers.
|
||
|
|
use_causal_conv (bool): Whether to use causal convolutions.
|
||
|
|
"""
|
||
|
|
cur_length = audio_length
|
||
|
|
|
||
|
|
if kernel_sizes is None or strides is None or dilations is None or use_causal_conv is None:
|
||
|
|
return cur_length
|
||
|
|
|
||
|
|
for kernel_size, stride, dilation in zip(kernel_sizes, strides, dilations):
|
||
|
|
effective_kernel_size = (kernel_size - 1) * dilation + 1
|
||
|
|
padding_total = kernel_size - stride
|
||
|
|
padding_right = padding_total // 2
|
||
|
|
padding_left = padding_total - padding_right
|
||
|
|
|
||
|
|
n_frames = (cur_length - effective_kernel_size + padding_total) / stride + 1
|
||
|
|
n_frames = math.ceil(n_frames) - 1
|
||
|
|
ideal_length = n_frames * stride + kernel_size - padding_total
|
||
|
|
extra_padding = ideal_length - cur_length
|
||
|
|
|
||
|
|
if use_causal_conv:
|
||
|
|
padding_left = padding_total
|
||
|
|
padding_right = extra_padding
|
||
|
|
else:
|
||
|
|
padding_right = padding_right + extra_padding
|
||
|
|
|
||
|
|
cur_length = cur_length + padding_left + padding_right
|
||
|
|
cur_length = (cur_length - dilation * (kernel_size - 1) - 1) // stride + 1
|
||
|
|
|
||
|
|
return cur_length
|
||
|
|
|
||
|
|
def save_audio(
|
||
|
|
self,
|
||
|
|
audio: AudioInput,
|
||
|
|
saving_path: str | Path | list[str | Path],
|
||
|
|
**kwargs: Unpack[CsmProcessorKwargs],
|
||
|
|
):
|
||
|
|
# TODO: @eustlb, this should be in AudioProcessor
|
||
|
|
if not is_soundfile_available():
|
||
|
|
raise ImportError("Please install `soundfile` to save audio files.")
|
||
|
|
|
||
|
|
# ensure correct audio input
|
||
|
|
audio = make_list_of_audio(audio)
|
||
|
|
|
||
|
|
# ensure correct saving path
|
||
|
|
if isinstance(saving_path, (str, Path)):
|
||
|
|
saving_path = [saving_path]
|
||
|
|
elif not (isinstance(saving_path, (list, tuple)) and all(isinstance(p, (str, Path)) for p in saving_path)):
|
||
|
|
raise ValueError("Invalid input path. Please provide a string, or a list of strings")
|
||
|
|
|
||
|
|
if len(audio) != len(saving_path):
|
||
|
|
raise ValueError("The number of audio and saving paths must be the same")
|
||
|
|
|
||
|
|
output_kwargs = self._merge_kwargs(
|
||
|
|
CsmProcessorKwargs,
|
||
|
|
**kwargs,
|
||
|
|
)
|
||
|
|
audio_kwargs = output_kwargs["audio_kwargs"]
|
||
|
|
sampling_rate = audio_kwargs["sampling_rate"]
|
||
|
|
|
||
|
|
for audio_value, p in zip(audio, saving_path):
|
||
|
|
if isinstance(audio_value, torch.Tensor):
|
||
|
|
audio_value = audio_value.cpu().float().numpy()
|
||
|
|
sf.write(p, audio_value, sampling_rate)
|
||
|
|
|
||
|
|
@auto_docstring
|
||
|
|
def __call__(
|
||
|
|
self,
|
||
|
|
text: TextInput | PreTokenizedInput | list[TextInput] | list[PreTokenizedInput] | None,
|
||
|
|
audio: AudioInput | None = None,
|
||
|
|
output_labels: bool | None = False,
|
||
|
|
depth_decoder_labels_ratio: float | None = 1.0,
|
||
|
|
**kwargs: Unpack[CsmProcessorKwargs],
|
||
|
|
):
|
||
|
|
r"""
|
||
|
|
output_labels (bool, *optional*, default=False):
|
||
|
|
Whether to return labels for training. Indices will be in `[config.audio_token_id, -100, -101]`.
|
||
|
|
- `config.audio_token_id` indicates an audio frame (considering sequence length elements as frames)
|
||
|
|
- `-100` will be ignored in the loss computation
|
||
|
|
- `-101` indicates the audio frame will be used only for the backbone model (using the first codebook token as labels)
|
||
|
|
depth_decoder_labels_ratio (float, *optional*, default=1.0):
|
||
|
|
The ratio of audio frames to keep for the depth decoder labels.
|
||
|
|
|
||
|
|
Returns:
|
||
|
|
[`BatchFeature`]: A [`BatchFeature`] with the following fields:
|
||
|
|
|
||
|
|
- **input_ids** -- List of token ids to be fed to a model. Returned when `text` is not `None`.
|
||
|
|
- **input_values** -- List of audio values to be fed to a model. Returned when `audio` is not `None`.
|
||
|
|
- **attention_mask** -- List of indices specifying which tokens should be attended to by the model (when
|
||
|
|
`return_attention_mask=True` or if *"attention_mask"* is in `self.model_input_names` and if `text` is not
|
||
|
|
`None`).
|
||
|
|
- **labels** -- List of labels for the audio frames. Returned when `output_labels=True`.
|
||
|
|
"""
|
||
|
|
|
||
|
|
output_kwargs = self._merge_kwargs(
|
||
|
|
CsmProcessorKwargs,
|
||
|
|
tokenizer_init_kwargs=self.tokenizer.init_kwargs,
|
||
|
|
**kwargs,
|
||
|
|
)
|
||
|
|
|
||
|
|
text_kwargs = output_kwargs["text_kwargs"]
|
||
|
|
audio_kwargs = output_kwargs["audio_kwargs"]
|
||
|
|
return_tensors = text_kwargs.get("return_tensors", None)
|
||
|
|
if return_tensors != "pt":
|
||
|
|
raise ValueError(f"{self.__class__.__name__} only supports `return_tensors='pt'`.")
|
||
|
|
|
||
|
|
if isinstance(text, str):
|
||
|
|
text = [text]
|
||
|
|
elif not (isinstance(text, (list, tuple)) and all(isinstance(t, str) for t in text)):
|
||
|
|
raise ValueError("Invalid input text. Please provide a string, or a list of strings")
|
||
|
|
n_audio_in_text = [t.count(self.audio_token) for t in text]
|
||
|
|
|
||
|
|
n_audio = 0
|
||
|
|
if audio is not None:
|
||
|
|
audio = make_list_of_audio(audio)
|
||
|
|
n_audio = len(audio)
|
||
|
|
|
||
|
|
if sum(n_audio_in_text) > 0 and n_audio != sum(n_audio_in_text):
|
||
|
|
if audio is None:
|
||
|
|
raise ValueError("No audio were provided, but there are audio tokens in the prompt")
|
||
|
|
else:
|
||
|
|
raise ValueError(
|
||
|
|
f"The number of audio tokens in each text ({n_audio_in_text}) should be the same as the "
|
||
|
|
f"number of provided audios ({n_audio})."
|
||
|
|
)
|
||
|
|
|
||
|
|
if audio is not None:
|
||
|
|
encoded_length_kwargs = audio_kwargs.pop("encoded_length_kwargs", {})
|
||
|
|
num_audio_tokens_list = [
|
||
|
|
self._get_encoded_length(audio_array.shape[-1], **encoded_length_kwargs) for audio_array in audio
|
||
|
|
]
|
||
|
|
num_audio_tokens_list_copy = num_audio_tokens_list.copy()
|
||
|
|
|
||
|
|
# expand the text to repeat the audio token for the corresponding number of frames
|
||
|
|
expanded_text = []
|
||
|
|
for sample in text:
|
||
|
|
replace_str = []
|
||
|
|
while self.audio_token in sample:
|
||
|
|
num_audio_tokens = num_audio_tokens_list_copy.pop(0)
|
||
|
|
expanded_audio_token = self.audio_token * num_audio_tokens
|
||
|
|
|
||
|
|
replace_str.append(expanded_audio_token)
|
||
|
|
sample = sample.replace(self.audio_token, "<placeholder>", 1)
|
||
|
|
|
||
|
|
while "<placeholder>" in sample:
|
||
|
|
sample = sample.replace("<placeholder>", replace_str.pop(0), 1)
|
||
|
|
expanded_text.append(sample)
|
||
|
|
|
||
|
|
text = expanded_text
|
||
|
|
|
||
|
|
encoding = self.tokenizer(text, **text_kwargs)
|
||
|
|
data = {}
|
||
|
|
data.update(encoding)
|
||
|
|
|
||
|
|
if audio is not None:
|
||
|
|
audio_kwargs.pop("return_attention_mask", None) # not supported by the feature extractor
|
||
|
|
|
||
|
|
concatenated_audio, input_values_cutoffs = [], []
|
||
|
|
offset = 0
|
||
|
|
for n_audio in n_audio_in_text:
|
||
|
|
if n_audio == 0:
|
||
|
|
concatenated_audio.append(np.zeros(0))
|
||
|
|
input_values_cutoffs.append(torch.tensor([-1]))
|
||
|
|
else:
|
||
|
|
concatenated_audio.append(
|
||
|
|
np.concatenate(
|
||
|
|
[
|
||
|
|
el.cpu().numpy() if isinstance(el, torch.Tensor) else el
|
||
|
|
for el in audio[offset : offset + n_audio]
|
||
|
|
],
|
||
|
|
axis=-1,
|
||
|
|
)
|
||
|
|
)
|
||
|
|
input_values_cutoffs.append(
|
||
|
|
torch.tensor([el.shape[-1] for el in audio[offset : offset + n_audio]]).cumsum(dim=-1)
|
||
|
|
)
|
||
|
|
offset += n_audio
|
||
|
|
|
||
|
|
audio_inputs = self.feature_extractor(concatenated_audio, **audio_kwargs)
|
||
|
|
audio_inputs.pop("padding_mask", None) # not applicable here
|
||
|
|
data.update(audio_inputs)
|
||
|
|
|
||
|
|
# pad and stack the audio cut idxs
|
||
|
|
max_len = max(cut_idxs.shape[-1] for cut_idxs in input_values_cutoffs)
|
||
|
|
input_values_cutoffs = [
|
||
|
|
torch.nn.functional.pad(cut_idxs, (0, max_len - cut_idxs.shape[-1]), value=-1)
|
||
|
|
for cut_idxs in input_values_cutoffs
|
||
|
|
]
|
||
|
|
data["input_values_cutoffs"] = torch.stack(input_values_cutoffs, dim=0)
|
||
|
|
|
||
|
|
if output_labels:
|
||
|
|
audio_frame_idxs = (data["input_ids"] == self.audio_token_id).nonzero()
|
||
|
|
n_audio_frames = audio_frame_idxs.shape[0]
|
||
|
|
|
||
|
|
if depth_decoder_labels_ratio <= 1.0:
|
||
|
|
rand_idxs = torch.randperm(n_audio_frames)[: int(n_audio_frames * (1 - depth_decoder_labels_ratio))]
|
||
|
|
skip_frames_idxs = audio_frame_idxs[rand_idxs]
|
||
|
|
else:
|
||
|
|
skip_frames_idxs = audio_frame_idxs
|
||
|
|
|
||
|
|
labels = torch.where(
|
||
|
|
(data["input_ids"] == self.audio_token_id) | (data["input_ids"] == self.audio_eos_token_id),
|
||
|
|
data["input_ids"],
|
||
|
|
-100,
|
||
|
|
)
|
||
|
|
labels[skip_frames_idxs[:, 0], skip_frames_idxs[:, 1]] = -101
|
||
|
|
|
||
|
|
data["labels"] = labels
|
||
|
|
|
||
|
|
return BatchFeature(data=data, tensor_type=return_tensors)
|
||
|
|
|
||
|
|
@property
|
||
|
|
def model_input_names(self):
|
||
|
|
tokenizer_input_names = self.tokenizer.model_input_names
|
||
|
|
feature_extractor_input_names = self.feature_extractor.model_input_names
|
||
|
|
|
||
|
|
# Remove `padding_mask`, it is popped and not used when processing. Make a copy of list when removing
|
||
|
|
# otherwise `self.feature_extractor.model_input_names` is also modified
|
||
|
|
feature_extractor_input_names = [name for name in feature_extractor_input_names if name != "padding_mask"]
|
||
|
|
return list(tokenizer_input_names + feature_extractor_input_names + ["input_values_cutoffs"])
|
||
|
|
|
||
|
|
|
||
|
|
__all__ = ["CsmProcessor"]
|