Any-to-Any
Transformers
Safetensors
lfm3_audio
automatic-speech-recognition
liquid
lfm3
audio
speech
edge
conversational
multilingual
custom_code
Instructions to use LiquidAI/LFM3-Audio-2B with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use LiquidAI/LFM3-Audio-2B with Transformers:
# pip install -U transformers accelerate # Load model directly from transformers import AutoModelForMultimodalLM model = AutoModelForMultimodalLM.from_pretrained("LiquidAI/LFM3-Audio-2B", trust_remote_code=True, device_map="auto") - Notebooks
- Google Colab
- Kaggle
File size: 18,723 Bytes
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#
# 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.
"""Processor class for LFM3-Audio."""
import copy
import math
import os
import re
from typing import TypedDict
import numpy as np
import torch
from transformers.audio_utils import AudioInput, make_list_of_audio
from transformers.configuration_utils import PreTrainedConfig
from transformers.feature_extraction_utils import BatchFeature
from transformers.processing_utils import ProcessingKwargs, ProcessorMixin
from transformers.utils import TensorType, auto_docstring
from .modeling_liquid_codec import LiquidCodecModel
# The prompt format of the training data: ChatML turns, one placeholder per clip, no thinking block
DEFAULT_CHAT_TEMPLATE = (
"{{- bos_token -}}"
"{%- for message in messages -%}"
"{{- '<|im_start|>' + message['role'] + '\\n' -}}"
"{%- if message['content'] is string -%}{{- message['content'] -}}"
"{%- else -%}{%- for content in message['content'] -%}"
"{%- if content['type'] == 'audio' -%}{{- '<|audio|>' -}}"
"{%- elif content['type'] == 'audio_out' -%}{{- '<|audio_out|>' -}}"
"{%- elif content['type'] == 'text' -%}{{- content['text'] -}}"
"{%- endif -%}{%- endfor -%}{%- endif -%}"
"{{- '<|im_end|>\\n' -}}"
"{%- endfor -%}"
"{%- if add_generation_prompt -%}{{- '<|im_start|>assistant\\n' -}}{%- endif -%}"
)
class Lfm3AudioFeatureExtractorKwargs(TypedDict, total=False):
"""
sampling_rate (`int`, *optional*):
The sampling rate of the clips, which must be the extractor's own; it is checked, never resampled.
return_tensors (`str` or [`~utils.TensorType`], *optional*):
The tensor type of the returned features.
load_audio_backend (`str`, *optional*):
The decoding backend `apply_chat_template` loads audio with; the extractor itself ignores it.
"""
# the extractor always right-pads to whole frames and returns the mask, so the padding, truncation and
# attention-mask options of `AudioKwargs` are not offered rather than silently dropped
sampling_rate: int | None
return_tensors: str | TensorType | None
load_audio_backend: str | None
class Lfm3AudioProcessorKwargs(ProcessingKwargs, total=False):
audio_kwargs: Lfm3AudioFeatureExtractorKwargs
_defaults = {}
@auto_docstring
class Lfm3AudioProcessor(ProcessorMixin):
r"""
Wraps the LFM3-Audio feature extractor, the LFM3 tokenizer and optionally the Liquid codec. The text carries one
`audio_token` per input clip and one `audio_out_token` per output clip or speech frame of the prompt; the
processor expands the clips to their encoder tokens (one per 8 spectrogram frames) and codec frames. The two
placeholder tokens are added to the tokenizer when it does not define them: the model replaces their embeddings
with the speech features, so they never reach the model as text.
"""
attributes = ["feature_extractor", "tokenizer"]
tokenizer_class = "AutoTokenizer"
audio_tokenizer_class = "LiquidCodecModel"
valid_processor_kwargs = Lfm3AudioProcessorKwargs
def __init__(
self,
feature_extractor=None,
tokenizer=None,
audio_tokenizer=None,
chat_template=None,
audio_token: str = "<|audio|>",
audio_out_token: str = "<|audio_out|>",
**kwargs,
):
r"""
feature_extractor ([`Lfm3AudioFeatureExtractor`]):
The feature extractor computing the log-mel spectrograms.
tokenizer ([`PreTrainedTokenizerFast`]):
The LFM3 tokenizer.
audio_tokenizer ([`LiquidCodecModel`], *optional*):
The codec turning 24 kHz output clips into speech frames.
chat_template (`str`, *optional*):
A Jinja template to turn conversations into prompts.
audio_token (`str`, *optional*, defaults to `"<|audio|>"`):
The placeholder token of one encoded frame of an input clip.
audio_out_token (`str`, *optional*, defaults to `"<|audio_out|>"`):
The placeholder token of one speech frame of the prompt.
"""
self.audio_token = getattr(tokenizer, "audio_token", audio_token)
self.audio_out_token = getattr(tokenizer, "audio_out_token", audio_out_token)
missing = [
token
for token in (self.audio_token, self.audio_out_token)
if tokenizer.convert_tokens_to_ids(token) in (None, tokenizer.unk_token_id)
]
if missing:
tokenizer.add_tokens(missing, special_tokens=True)
self.audio_token_id = tokenizer.convert_tokens_to_ids(self.audio_token)
self.audio_out_token_id = tokenizer.convert_tokens_to_ids(self.audio_out_token)
if chat_template is None:
chat_template = DEFAULT_CHAT_TEMPLATE
super().__init__(feature_extractor, tokenizer, chat_template=chat_template)
# a released transformers only accepts its own codec classes as `audio_tokenizer`
self.audio_tokenizer = audio_tokenizer
@classmethod
def from_pretrained(cls, pretrained_model_name_or_path, *args, audio_tokenizer=None, **kwargs):
# `ProcessorMixin` forwards unknown kwargs to the tokenizer loader, which would keep the codec object in the
# tokenizer config and break `save_pretrained`
processor = super().from_pretrained(pretrained_model_name_or_path, *args, **kwargs)
if audio_tokenizer is not None:
processor.audio_tokenizer = audio_tokenizer
return processor
def to_dict(self):
# `ProcessorMixin.to_dict` deep-copies the attributes; the codec weights are saved on their own. A shallow
# clone drops the codec, so concurrent calls on `self` keep it.
clone = copy.copy(self)
codec = clone.__dict__.pop("audio_tokenizer", None)
output = ProcessorMixin.to_dict(clone)
if codec is not None:
output["audio_tokenizer_subfolder"] = "audio_tokenizer"
return output
def save_pretrained(self, save_directory, push_to_hub=False, **kwargs):
files = super().save_pretrained(save_directory, push_to_hub=False, **kwargs)
if getattr(self, "audio_tokenizer", None) is not None:
self.audio_tokenizer.save_pretrained(os.path.join(save_directory, "audio_tokenizer"))
if push_to_hub:
repo_id = kwargs.pop("repo_id", os.path.basename(os.path.abspath(save_directory)))
self.push_to_hub(repo_id, **kwargs)
return files
@classmethod
def get_processor_dict(cls, pretrained_model_name_or_path, **kwargs):
codec_kwargs = {
key: value
for key, value in kwargs.items()
if key in ("cache_dir", "force_download", "local_files_only", "token", "revision")
}
parent = kwargs.get("subfolder") or ""
processor_dict, kwargs = super().get_processor_dict(pretrained_model_name_or_path, **kwargs)
if "processor_class" not in processor_dict:
config, _ = PreTrainedConfig.get_config_dict(
pretrained_model_name_or_path, subfolder=parent, **codec_kwargs
)
if config.get("model_type") == "lfm3_audio" and config.get("format_version") == 1:
# Legacy exports carry the text-only tokenizer template, not the trained audio prompt format.
processor_dict["chat_template"] = DEFAULT_CHAT_TEMPLATE
subfolder = processor_dict.pop("audio_tokenizer_subfolder", None)
if subfolder is not None:
if subfolder != "audio_tokenizer":
raise ValueError("Expected the bundled codec in the audio_tokenizer subfolder")
codec_class = LiquidCodecModel
processor_dict["audio_tokenizer"] = codec_class.from_pretrained(
pretrained_model_name_or_path, subfolder=os.path.join(parent, subfolder), **codec_kwargs
)
return processor_dict, kwargs
@auto_docstring
def __call__(self, text=None, audio: AudioInput | None = None, audio_out: AudioInput | None = None, **kwargs):
r"""
audio_out (`np.ndarray`, `torch.Tensor` or a list of them, *optional*):
24 kHz clips, one per `audio_out_token` in the text, encoded with the codec into `audio_out_codes`; the
placeholder is repeated once per frame plus once for the end-of-speech frame that closes every clip.
"""
# the shared audio loader indexes the first clip before `validate_inputs` runs
for name, clips in (("audio", audio), ("audio_out", audio_out)):
if isinstance(clips, (list, tuple)) and len(clips) == 0:
raise ValueError(f"`{name}` must hold at least one clip; pass `{name}=None` for prompts without one")
if audio_out is None:
return super().__call__(text=text, audio=audio, **kwargs)
if text is None:
raise ValueError("`audio_out` needs the text holding its placeholders")
single = isinstance(text, str)
prompts = [text] if single else list(text)
expanded, codes, audio_out_offsets = self._expand_audio_out(prompts, audio_out)
merged = self._merge_kwargs(
self.valid_processor_kwargs, tokenizer_init_kwargs=self.tokenizer.init_kwargs, **kwargs
)
return_tensors = merged["text_kwargs"].get("return_tensors")
outputs = super().__call__(text=expanded[0] if single else expanded, audio=audio, **kwargs)
self._check_audio_out_placeholders(outputs["input_ids"], codes.shape[0])
if "text_replacement_offsets" in outputs:
# the shared expansion saw the text after the speech placeholders were repeated
outputs["text_replacement_offsets"] = self._compose_replacement_offsets(
prompts, audio_out_offsets, outputs["text_replacement_offsets"]
)
outputs["audio_out_codes"] = self._as_return_tensors(codes, return_tensors)
return outputs
def _check_audio_out_placeholders(self, input_ids, num_frames: int) -> None:
# truncation may have dropped placeholders the codes still cover
rows = input_ids.tolist() if hasattr(input_ids, "tolist") else input_ids
rows = [rows] if rows and isinstance(rows[0], int) else rows
num_placeholders = sum(row.count(self.audio_out_token_id) for row in rows)
if num_placeholders != num_frames:
raise ValueError(
f"Found {num_placeholders} {self.audio_out_token} tokens after tokenization for {num_frames} "
"speech frames; disable truncation or shorten the prompt"
)
@staticmethod
def _as_return_tensors(codes: torch.Tensor, return_tensors):
return BatchFeature({"codes": codes}, tensor_type=return_tensors)["codes"]
@staticmethod
def _rewrite_with_offsets(prompt: str, pattern: str, replace, kind: str) -> tuple[str, list[dict]]:
"""Rewrites `prompt` like `ProcessorMixin.get_text_with_replacements`, with the same offset records."""
pieces, row, last, offset = [], [], 0, 0
for match in re.finditer(pattern, prompt):
start, end = match.span()
replacement = replace(match)
pieces.extend((prompt[last:start], replacement))
row.append(
{
"type": kind,
"span": (start, end),
"new_span": (start + offset, start + offset + len(replacement)),
"text": match.group(),
"replacement": replacement,
}
)
offset += len(replacement) - (end - start)
last = end
pieces.append(prompt[last:])
return "".join(pieces), row
@staticmethod
def _compose_replacement_offsets(prompts: list[str], first: list[list[dict]], second) -> list[list[dict]]:
"""The records of two successive rewrites (the caller's text A to B, then B to C) as one rewrite of A: an entry
of the second that covers entries of the first absorbs them, so every entry spans the caller's text."""
second = second or [[] for _ in prompts]
composed = []
for prompt, row_first, row_second in zip(prompts, first, second):
def to_first(position, row_first=row_first):
return position - sum(
len(r["replacement"]) - (r["span"][1] - r["span"][0])
for r in row_first
if r["new_span"][1] <= position
)
mapped = []
for entry in row_second:
start, end = to_first(entry["span"][0]), to_first(entry["span"][1])
mapped.append({**entry, "span": (start, end), "text": prompt[start:end]})
kept = [
r
for r in row_first
if not any(m["span"][0] <= r["span"][0] and r["span"][1] <= m["span"][1] for m in mapped)
]
row, offset = [], 0
for entry in sorted(kept + mapped, key=lambda r: r["span"]):
start, end = entry["span"]
row.append({**entry, "new_span": (start + offset, start + offset + len(entry["replacement"]))})
offset += len(entry["replacement"]) - (end - start)
composed.append(row)
return composed
def _expand_audio_out(
self, prompts: list[str], audio_out: AudioInput
) -> tuple[list[str], torch.Tensor, list[list[dict]]]:
if getattr(self, "audio_tokenizer", None) is None:
raise ValueError("`audio_out` requires the processor to hold the codec as `audio_tokenizer`")
clips = make_list_of_audio(audio_out)
if sum(prompt.count(self.audio_out_token) for prompt in prompts) != len(clips):
raise ValueError(f"The number of {self.audio_out_token} tokens in the text must match the number of clips")
codes = [self.encode_audio(clip) for clip in clips]
# the model ends speech with a frame of end-of-speech codes before the next text token; the prompt and the
# training targets carry it the same way `generate(return_audio_eos=True)` does
eos_frame = torch.full((1, codes[0].shape[1]), self.audio_eos_code, dtype=codes[0].dtype)
codes = [torch.cat((frames, eos_frame)) for frames in codes]
frame_counts = iter(code.shape[0] for code in codes)
expanded, offsets = [], []
for prompt in prompts:
prompt, row = self._rewrite_with_offsets(
prompt,
re.escape(self.audio_out_token),
lambda match: self.audio_out_token * next(frame_counts),
"audio_out",
)
expanded.append(prompt)
offsets.append(row)
return expanded, torch.cat(codes), offsets
@property
def audio_eos_code(self) -> int:
"""The end-of-speech code: the first id after the codec's vocabulary."""
return math.prod(self.audio_tokenizer.config.fsq_levels)
@torch.no_grad()
def encode_audio(self, audio) -> torch.Tensor:
codec = self.audio_tokenizer
waveform = torch.as_tensor(audio if torch.is_tensor(audio) else np.asarray(audio), dtype=torch.float32)
# the quantizer rounds non-finite values to INT64_MIN, which would surface as codes
if waveform.ndim != 1 or not torch.isfinite(waveform).all():
raise ValueError("`audio_out` clips must be mono waveforms of finite samples")
# an offloaded codec keeps some weights on `meta`: send the clip to a device that holds weights, and the hooks
# move it on from there
device = next(
(param.device for param in codec.parameters() if param.device.type != "meta"), torch.device("cpu")
)
# the codec's return type follows its config's `return_dict`
codes = codec.encode(waveform.to(device)[None], return_dict=True).audio_codes[0].T.cpu()
# a compiled codec skips its latent check, and a non-finite latent rounds to INT64_MIN
if codes.numel() and not 0 <= int(codes.min()) <= int(codes.max()) < self.audio_eos_code:
raise ValueError("The audio overflows the codec; expected `audio_out` samples in [-1, 1]")
return codes
def validate_inputs(self, images=None, text=None, videos=None, audio=None, **kwargs):
super().validate_inputs(images=images, text=text, videos=videos, audio=audio, **kwargs)
if text is None:
return
prompts = [text] if isinstance(text, str) else text
num_placeholders = sum(prompt.count(self.audio_token) for prompt in prompts)
num_clips = 0 if audio is None else len(make_list_of_audio(audio))
if num_placeholders != num_clips:
raise ValueError(
f"Found {num_placeholders} {self.audio_token} tokens in the text but received {num_clips} clips"
)
def _process_audio(self, audio: AudioInput, **kwargs):
# the backend only serves `apply_chat_template`'s audio loading
kwargs.pop("load_audio_backend", None)
audio_inputs = self.feature_extractor(audio, **kwargs)
lengths = np.asarray(audio_inputs["feature_attention_mask"]).sum(-1).tolist()
audio_inputs["num_audio_tokens"] = [
self.feature_extractor.num_encoder_tokens(int(length)) for length in lengths
]
audio_replacements = [self.replace_audio_token(audio_inputs, audio_idx=idx) for idx in range(len(lengths))]
return audio_inputs, audio_replacements
def replace_audio_token(self, audio_inputs: dict, audio_idx: int, **kwargs) -> str:
return self.audio_token * audio_inputs["num_audio_tokens"][audio_idx]
@property
def unused_input_names(self) -> list[str]:
return ["num_audio_tokens"]
@property
def model_input_names(self):
tokenizer_input_names = self.tokenizer.model_input_names
feature_extractor_input_names = self.feature_extractor.model_input_names
return list(dict.fromkeys(tokenizer_input_names + feature_extractor_input_names + ["audio_out_codes"]))
__all__ = ["Lfm3AudioProcessor"]
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