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1092 lines (971 loc) · 45.2 KB
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import logging
import os
import random
import re
import string
import time
import traceback
from typing import Union
import torch
import torch.nn as nn
from funasr.metrics.compute_acc import compute_accuracy
from funasr.register import tables
from funasr.train_utils.device_funcs import force_gatherable, to_device
from funasr.utils.datadir_writer import DatadirWriter
from funasr.utils.load_utils import extract_fbank, load_audio_text_image_video
try:
from transformers import AutoConfig, AutoModelForCausalLM
except ImportError:
AutoConfig = None
AutoModelForCausalLM = None
from .ctc import CTC
from .checkpoint_utils import disable_incomplete_ctc, normalize_checkpoint_state
from .device_utils import resolve_autocast_device_type
from .tools.utils import forced_align
dtype_map = {"bf16": torch.bfloat16, "fp16": torch.float16, "fp32": torch.float32}
@tables.register("model_classes", "FunASRNano")
class FunASRNano(nn.Module):
"""Fun-ASR-Nano: End-to-End ASR Large Model.
Trained on tens of millions of hours of real speech data.
Language coverage is checkpoint-specific: Nano supports Chinese, English,
and Japanese plus Chinese dialects/accents; MLT-Nano supports 31 languages.
Features:
- Character-level timestamps (via CTC forced alignment)
- Hotword customization
- Speaker diarization (when combined with spk_model)
- Lyrics and rap recognition
- Streaming chunk-by-chunk inference (demo2.py)
Output: {"key": ..., "text": ..., "timestamps": [{"token", "start_time", "end_time"}, ...],
"ctc_timestamps": [...]}
Note: Outputs punctuation natively — punc_model is NOT needed.
Requirements: pip install tiktoken huggingface_hub
"""
def __init__(
self,
audio_encoder: str = None,
audio_encoder_conf: dict = None,
audio_adaptor: str = None,
audio_adaptor_conf: dict = None,
llm: str = None,
llm_conf: dict = None,
input_size: int = 80,
length_normalized_loss: bool = False,
**kwargs,
):
"""Initialize FunASRNano.
Args:
audio_encoder: TODO.
audio_encoder_conf: Configuration dict for audio_encoder.
audio_adaptor: TODO.
audio_adaptor_conf: Configuration dict for audio_adaptor.
llm: TODO.
llm_conf: Configuration dict for llm.
input_size: Size/dimension parameter.
length_normalized_loss: TODO.
**kwargs: Additional keyword arguments.
"""
super().__init__()
# audio encoder
hub = audio_encoder_conf.get("hub", None)
self.audio_encoder_activation_checkpoint = audio_encoder_conf.get(
"activation_checkpoint", False
)
if hub == "ms":
from funasr import AutoModel
model = AutoModel(model=audio_encoder, model_revision="master")
audio_encoder_output_size = (
model.model.encoder_output_size
if hasattr(model.model, "encoder_output_size")
else -1
)
audio_encoder = (
model.model.model.encoder if hasattr(model.model, "model") else model.model.encoder
)
else:
encoder_class = tables.encoder_classes.get(audio_encoder)
audio_encoder = encoder_class(input_size=input_size, **audio_encoder_conf)
audio_encoder_output_size = audio_encoder.output_size()
freeze = audio_encoder_conf.get("freeze", True)
if freeze:
for _, param in audio_encoder.named_parameters():
param.requires_grad = False
audio_encoder.eval()
self.audio_encoder = audio_encoder
# llm
self.llm = None
init_param_path = llm_conf.get("init_param_path", None)
llm_dim = None
llm_load_kwargs = llm_conf.get("load_kwargs", {})
config = AutoConfig.from_pretrained(init_param_path)
model = AutoModelForCausalLM.from_config(config, **llm_load_kwargs)
freeze = llm_conf.get("freeze", True)
if freeze:
for _, param in model.named_parameters():
param.requires_grad = False
model.eval()
if llm_conf.get("activation_checkpoint", False):
model.gradient_checkpointing_enable()
self.llm_dtype = llm_conf.get("llm_dtype", "fp32")
self.llm = model.to(dtype_map[self.llm_dtype])
llm_dim = model.get_input_embeddings().weight.shape[-1]
# lora: inject LoRA adapters into the LLM target Linear layers
if self.llm is not None and llm_conf.get("use_lora", False):
self._apply_lora_to_llm(llm_conf)
# adaptor
adaptor_class = tables.adaptor_classes.get(audio_adaptor)
if audio_encoder_output_size > 0:
audio_adaptor_conf["encoder_dim"] = audio_encoder_output_size
audio_adaptor_conf["llm_dim"] = (
llm_dim if llm_dim is not None else audio_adaptor_conf["llm_dim"]
)
audio_adaptor = adaptor_class(**audio_adaptor_conf)
freeze = audio_adaptor_conf.get("freeze", False)
if freeze:
for _, param in audio_adaptor.named_parameters():
param.requires_grad = False
audio_adaptor.eval()
self.audio_adaptor = audio_adaptor
self.use_low_frame_rate = audio_adaptor_conf.get("use_low_frame_rate", False)
# ctc decoder
self.ctc_decoder = None
self._externally_loaded_ctc_keys = set()
# TODO: fix table name
ctc_decoder_class = tables.adaptor_classes.get(kwargs.get("ctc_decoder", None))
if ctc_decoder_class is not None:
ctc_tokenizer = (
kwargs.get("ctc_tokenizer", None)
if "ctc_tokenizer" in kwargs
else kwargs["dataset_conf"]["ctc_tokenizer"]
)
ctc_tokenizer_conf = (
kwargs.get("ctc_tokenizer_conf", None)
if "ctc_tokenizer_conf" in kwargs
else kwargs["dataset_conf"]["ctc_tokenizer_conf"]
)
if ctc_tokenizer is not None and ctc_tokenizer_conf is not None:
ctc_tokenizer_class = tables.tokenizer_classes.get(ctc_tokenizer)
ctc_tokenizer = ctc_tokenizer_class(**ctc_tokenizer_conf)
self.ctc_tokenizer = ctc_tokenizer
assert ctc_tokenizer is not None, f"ctc_tokenizer must be set"
ctc_vocab_size = kwargs.get("ctc_vocab_size", 60515)
ctc_decoder_conf = kwargs.get("ctc_decoder_conf", {})
if audio_encoder_output_size > 0:
ctc_decoder_conf["encoder_dim"] = audio_encoder_output_size
self.ctc_decoder = ctc_decoder_class(**ctc_decoder_conf)
init_param_path = ctc_decoder_conf.get("init_param_path", None)
if init_param_path is not None:
src_state = normalize_checkpoint_state(
torch.load(init_param_path, map_location="cpu")
)
flag = self.ctc_decoder.load_state_dict(src_state, strict=False)
self._externally_loaded_ctc_keys.update(
f"ctc_decoder.{key}"
for key in self.ctc_decoder.state_dict()
if key in src_state
)
logging.info(f"Loading ctc_decoder ckpt: {init_param_path}, status: {flag}")
freeze = ctc_decoder_conf.get("freeze", False)
if freeze:
for _, param in self.ctc_decoder.named_parameters():
param.requires_grad = False
self.ctc_decoder.eval()
ctc_conf = kwargs.get("ctc_conf", {})
self.blank_id = ctc_conf.get("blank_id", ctc_vocab_size - 1)
self.ctc_weight = kwargs.get("ctc_weight", 0.3)
self.ctc = CTC(
odim=ctc_vocab_size,
encoder_output_size=audio_encoder_output_size,
blank_id=self.blank_id,
**ctc_conf,
)
self.detach_ctc_decoder = kwargs.get("detach_ctc_decoder", True)
self.error_calculator = None
self.length_normalized_loss = length_normalized_loss
rank = int(os.environ.get("RANK", 0))
logging.info(f"rank: {rank}, model is builded.")
def _apply_lora_to_llm(self, llm_conf: dict):
"""Replace the LLM target Linear layers with LoRA adapters.
When ``llm_conf.use_lora`` is true, every ``nn.Linear`` in the LLM whose
module name contains one of ``lora_conf.target_modules`` is swapped for a
``lora.Linear`` (base weight shared and frozen, trainable ``lora_A`` /
``lora_B`` added). The base weights stay untouched in the state dict, so a
LoRA checkpoint can be loaded back into a model built with the same
``lora_conf``, or folded for deployment (W' = W + alpha/r * B @ A).
Frozen-base behaviour: the LLM is frozen by ``llm_conf.freeze``; the
``lora_A``/``lora_B`` parameters created here are trainable regardless.
With ``lora_only: true`` in the training config, ``mark_only_lora_as_trainable``
additionally freezes every non-LoRA parameter (encoder/adaptor/CTC), giving
pure-LoRA training. To keep the encoder/adaptor trainable while LoRA-tweaking
only the LLM, set ``lora_only: false`` and unfreeze them via their conf.
"""
lora_conf = llm_conf.get("lora_conf", {})
lora_r = lora_conf.get("r", 16)
lora_alpha = lora_conf.get("lora_alpha", 32)
lora_dropout = lora_conf.get("lora_dropout", 0.05)
target_modules = lora_conf.get("target_modules", ["q_proj", "v_proj"])
from funasr.models.lora.layers import Linear as LoRALinear
lora_applied = 0
for name, module in list(self.llm.named_modules()):
if not isinstance(module, nn.Linear):
continue
if not any(target in name.split(".") for target in target_modules):
continue
parts = name.split(".")
parent = self.llm
for p in parts[:-1]:
parent = getattr(parent, p)
new_linear = LoRALinear(
in_features=module.in_features,
out_features=module.out_features,
r=lora_r,
lora_alpha=lora_alpha,
lora_dropout=lora_dropout,
bias=module.bias is not None,
# keep the adapter params in the base weight's dtype (e.g. bf16),
# so the LoRA path does not depend on autocast to reconcile dtypes
dtype=module.weight.dtype,
)
# share (and keep frozen) the pretrained base weight
new_linear.weight = module.weight
new_linear.bias = module.bias
setattr(parent, parts[-1], new_linear)
lora_applied += 1
if lora_applied > 0:
logging.info(
"LoRA applied to %d Linear layers in the LLM "
"(r=%d, alpha=%d, dropout=%s, targets=%s)",
lora_applied,
lora_r,
lora_alpha,
lora_dropout,
target_modules,
)
else:
logging.warning(
"use_lora=true but no target modules found in the LLM "
"(target_modules=%s)",
target_modules,
)
def on_pretrained_model_loaded(self, loaded_keys):
"""Fail closed when a checkpoint configures CTC without trained weights."""
loaded_keys = set(loaded_keys).union(getattr(self, "_externally_loaded_ctc_keys", ()))
disable_incomplete_ctc(self, loaded_keys, log=logging)
def forward(
self,
speech: torch.Tensor = None,
speech_lengths: torch.Tensor = None,
input_ids: torch.Tensor = None,
attention_mask: torch.Tensor = None,
labels_ids: torch.Tensor = None,
fbank_beg: torch.Tensor = None,
fbank_mask: torch.Tensor = None,
**kwargs,
):
"""Forward pass for training.
Args:
speech: Speech audio tensor, shape (batch, time).
speech_lengths: Length of each speech sample.
input_ids: TODO.
attention_mask: TODO.
labels_ids: TODO.
fbank_beg: TODO.
fbank_mask: TODO.
**kwargs: Additional keyword arguments.
"""
batch_size, token_num = input_ids.shape
stats = {}
input_ids[input_ids < 0] = 0
inputs_embeds = self.llm.model.get_input_embeddings()(input_ids)
if speech is not None:
if len(speech_lengths.size()) > 1:
speech_lengths = speech_lengths[:, 0]
batch_size_speech, frames, _ = speech.shape
# audio encoder
if self.audio_encoder_activation_checkpoint:
from torch.utils.checkpoint import checkpoint
encoder_out, encoder_out_lens = checkpoint(
self.encode, speech, speech_lengths, use_reentrant=False
)
else:
encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
# audio_adaptor
encoder_out, encoder_out_lens = self.audio_adaptor(encoder_out, encoder_out_lens)
batch_size, token_num, dims = inputs_embeds.shape
fake_token_len = kwargs.get("fake_token_len")
fake_token_len[fake_token_len < 0] = 0
fbank_beg[fbank_beg < 0] = 0
speech_idx = 0
for batch_idx in range(batch_size):
for turn_id in range(fbank_beg.shape[1]):
fbank_beg_idx = fbank_beg[batch_idx, turn_id].item()
if fbank_beg_idx > 0:
speech_token_len = fake_token_len[batch_idx, turn_id]
speech_token = encoder_out[speech_idx, :speech_token_len, :]
try:
inputs_embeds[
batch_idx,
fbank_beg_idx : fbank_beg_idx + speech_token_len,
:,
] = speech_token
except Exception as e:
logging.error(f"{str(e)}, {traceback.format_exc()}")
logging.info(
f"batch_idx: {batch_idx}, inputs_embeds: {inputs_embeds.shape}, fbank_beg_idx: {fbank_beg_idx}, speech_token_len: {speech_token_len}, encoder_out: {encoder_out.shape}, encoder_out_lens: {encoder_out_lens}, fake_token_len: {fake_token_len}, speech_lengths: {speech_lengths}"
)
speech_token_len = encoder_out_lens[speech_idx].item()
speech_token = encoder_out[speech_idx, :speech_token_len, :]
inputs_embeds[
batch_idx,
fbank_beg_idx : fbank_beg_idx + speech_token_len,
:,
] = speech_token
speech_idx += 1
stats["batch_size_speech"] = batch_size_speech
stats["batch_size_x_frames"] = frames * batch_size_speech
stats["batch_size_real_frames"] = speech_lengths.sum().item()
stats["padding_frames"] = stats["batch_size_x_frames"] - stats["batch_size_real_frames"]
autocast_device_type = resolve_autocast_device_type(next(self.parameters()).device)
with torch.autocast(
device_type=autocast_device_type,
enabled=True if self.llm_dtype != "fp32" else False,
dtype=dtype_map[self.llm_dtype],
):
labels_ids[labels_ids == -1] = -100
attention_mask[attention_mask < 0] = 0
model_outputs = self.llm(
inputs_embeds=inputs_embeds.to(dtype_map[self.llm_dtype]),
attention_mask=attention_mask,
labels=labels_ids,
)
loss = model_outputs.loss
with torch.no_grad():
preds = torch.argmax(model_outputs.logits, -1)
acc_att = compute_accuracy(preds[:, :-1], labels_ids[:, 1:], ignore_label=-100)
stats["acc"] = acc_att
stats["loss"] = torch.clone(loss.detach())
stats["batch_size"] = batch_size
stats["batch_size_x_tokens"] = token_num * batch_size
stats["batch_size_real_tokens"] = attention_mask.sum().item()
stats["padding_tokens"] = stats["batch_size_x_tokens"] - stats["batch_size_real_tokens"]
dialog_turns = (fbank_beg > 0).sum(-1)
dialog_turns_max = torch.max(dialog_turns).int().item()
dialog_turns_avg = dialog_turns.sum().item() / batch_size
stats["dialog_turns_max"] = dialog_turns_max
stats["dialog_turns_avg"] = dialog_turns_avg
# force_gatherable: to-device and to-tensor if scalar for DataParallel
if self.length_normalized_loss:
batch_size = int((labels_ids > 0 + 1).sum())
loss, stats, weight = force_gatherable((loss, stats, batch_size), loss.device)
return loss, stats, weight
def forward_export(self, speech, speech_lengths, **kwargs):
"""Forward export.
Args:
speech: Speech audio tensor, shape (batch, time).
speech_lengths: Length of each speech sample.
**kwargs: Additional keyword arguments.
"""
x, olens = self.audio_encoder(speech, speech_lengths)
encoder_out, encoder_out_lens = self.audio_adaptor(x, olens)
return encoder_out, encoder_out_lens
def encode(self, speech, speech_lengths):
# audio encoder
"""Encode.
Args:
speech: Speech audio tensor, shape (batch, time).
speech_lengths: Length of each speech sample.
"""
encoder_out, encoder_out_lens = self.audio_encoder(speech, speech_lengths)
return encoder_out, encoder_out_lens
def data_template(self, data):
"""Data template.
Args:
data: TODO.
"""
system, user, assistant = [], [], []
for i, item in enumerate(data):
role = item["role"]
content = item["content"]
if role == "system":
system.append(content)
elif role == "user":
if "audio" in item:
audio = item["audio"]
content = [content, audio]
user.append(content)
elif role == "assistant":
assistant.append(content)
system = system * len(user)
contents = {
"system": system,
"user": user,
"assistant": assistant,
}
return contents
def data_load_speech(self, contents: dict, tokenizer, frontend, meta_data={}, **kwargs):
"""Data load speech.
Args:
contents: TODO.
tokenizer: Tokenizer instance for text encoding/decoding.
frontend: Audio frontend for feature extraction.
meta_data: TODO.
**kwargs: Additional keyword arguments.
"""
system = contents["system"]
user = contents["user"]
assistant = contents["assistant"]
pattern = re.compile(r"(<\|startofspeech\|>.*?<\|endofspeech\|>)")
do_think = True
sys_prompt = True
if "dataset_conf" in kwargs:
do_think = kwargs["dataset_conf"].get("do_think", True)
sys_prompt = kwargs["dataset_conf"].get("sys_prompt", True)
input_ids, labels, fbank, fbank_lens, fbank_mask, fbank_beg, fake_token_len = (
[],
[],
[],
[],
[],
[],
[],
)
input_source_ids = []
for i, (system_prompt, user_prompt, target_out) in enumerate(zip(system, user, assistant)):
if i >= kwargs.get("multiturn_num_max", 5):
break
if len(input_ids) > kwargs.get("max_token_length", 1500):
break
if isinstance(user_prompt, (list, tuple)):
user_prompt, audio = user_prompt
if i == 0:
if kwargs.get("infer_with_assistant_input", False):
source_input = f"<|im_start|>system\n{system_prompt}<|im_end|>\n<|im_start|>user\n{user_prompt}"
if not sys_prompt:
source_input = f"<|im_start|>user\n{user_prompt}"
else:
source_input = f"<|im_start|>system\n{system_prompt}<|im_end|>\n<|im_start|>user\n{user_prompt}<|im_end|>\n<|im_start|>assistant\n"
if not sys_prompt:
source_input = (
f"<|im_start|>user\n{user_prompt}<|im_end|>\n<|im_start|>assistant\n"
)
else:
if kwargs.get("infer_with_assistant_input", False):
source_input = f"<|im_start|>user\n{user_prompt}"
else:
source_input = (
f"<|im_start|>user\n{user_prompt}<|im_end|>\n<|im_start|>assistant\n"
)
if not do_think:
source_input += "<think>\n\n</think>\n\n"
if kwargs.get("prev_text", None) is not None:
source_input += kwargs["prev_text"]
splits = pattern.split(source_input)
source_ids = []
fbank_mask_i = []
fake_token_len_i = 0
fbank_beg_i = -1
speech, speech_lengths = [], []
for k, sub_str in enumerate(splits):
if not sub_str.startswith("<|startofspeech|>"):
sub_token = tokenizer.encode(sub_str)
source_ids += sub_token
fbank_mask_i += [0] * len(sub_token)
else:
sub_str = sub_str.replace("<|startofspeech|>", "").replace(
"<|endofspeech|>", ""
)
if sub_str.startswith("!"):
sub_str = sub_str[1:]
if sub_str.startswith("!"): # !!: audio sample point
sub_str = audio
try:
time1 = time.perf_counter()
data_src = load_audio_text_image_video(
sub_str, fs=frontend.fs, **kwargs
)
time2 = time.perf_counter()
meta_data["load_data"] = f"{time2 - time1:0.3f}"
except Exception as e:
logging.error(f"Loading wav failed! {str(e)}, {traceback.format_exc()}")
speech, speech_lengths = extract_fbank(
data_src,
data_type=kwargs.get("data_type", "sound"),
frontend=frontend,
is_final=True,
) # speech: [b, T, d]
time3 = time.perf_counter()
meta_data["extract_feat"] = f"{time3 - time2:0.3f}"
meta_data["batch_data_time"] = (
speech_lengths.sum().item()
* frontend.frame_shift
* frontend.lfr_n
/ 1000
)
if self.use_low_frame_rate:
olens = 1 + (speech_lengths[0].item() - 3 + 2 * 1) // 2
olens = 1 + (olens - 3 + 2 * 1) // 2
fake_token_len_i = (olens - 1) // 2 + 1
else:
fake_token_len_i = speech_lengths[0].item()
fake_token = [0] * fake_token_len_i
fbank_beg_i = len(source_ids)
source_ids += fake_token
fbank_mask_i += [1] * len(fake_token)
fbank_beg += [fbank_beg_i + len(input_ids)]
fake_token_len += [fake_token_len_i]
source_mask = [-100] * len(source_ids)
target_out = f"{target_out}<|im_end|>"
target_ids = tokenizer.encode(target_out)
input_source_ids = input_ids + source_ids
input_ids += source_ids + target_ids
labels += source_mask + target_ids
fbank_mask += fbank_mask_i
if len(speech) > 0:
fbank.append(speech[0, :, :])
fbank_lens.append(speech_lengths)
input_ids = torch.tensor(input_ids, dtype=torch.int64) # [: self.max_token_length]
attention_mask = torch.tensor([1] * len(input_ids), dtype=torch.int32)
labels = torch.tensor(labels, dtype=torch.int64) # [: self.max_token_length]
fbank_mask = torch.tensor(fbank_mask, dtype=torch.float32)
fbank_beg = torch.tensor(fbank_beg, dtype=torch.int32)
fake_token_len = torch.tensor(fake_token_len, dtype=torch.int32)
source_ids = torch.tensor(input_source_ids, dtype=torch.int64)
target_ids = torch.tensor(target_ids, dtype=torch.int64)
if len(fbank) > 0:
speech = torch.nn.utils.rnn.pad_sequence(fbank, batch_first=True, padding_value=0.0)
speech_lengths = torch.nn.utils.rnn.pad_sequence(
fbank_lens, batch_first=True, padding_value=-1
)
else:
speech = []
speech_lengths = []
output = {
"speech": speech,
"speech_lengths": speech_lengths,
"fbank_mask": fbank_mask[None, :],
"fbank_beg": fbank_beg[None,],
"fake_token_len": fake_token_len[None, :],
"input_ids": input_ids[None,],
"attention_mask": attention_mask[None,],
"labels_ids": labels,
"source_ids": source_ids[None, :],
"target_ids": target_ids[None, :],
}
return output
def inference_prepare(
self,
data_in,
data_lengths=None,
key: list = None,
tokenizer=None,
frontend=None,
**kwargs,
):
"""Inference prepare.
Args:
data_in: Input data (audio samples, file paths, or text).
data_lengths: Lengths of each input sample in the batch.
key: Sample identifiers.
tokenizer: Tokenizer instance for text encoding/decoding.
frontend: Audio frontend for feature extraction.
**kwargs: Additional keyword arguments.
"""
meta_data = {}
if len(data_in) > 1:
raise NotImplementedError("batch decoding is not implemented")
contents = self.data_template(data_in[0])
output = self.data_load_speech(contents, tokenizer, frontend, meta_data=meta_data, **kwargs)
batch = to_device(output, kwargs["device"])
# audio encoder
speech = batch["speech"]
if len(speech) > 0:
if "audio_embedding" in kwargs and "audio_embedding_lens" in kwargs:
encoder_out = kwargs["audio_embedding"]
encoder_out_lens = kwargs["audio_embedding_lens"]
else:
speech_lengths = batch["speech_lengths"][:, 0]
# NOTE: the audio encoder contains fp32-only ops, so casting its
# input to fp16/bf16 here raises a dtype mismatch. The encoder
# therefore always runs in fp32; low precision (fp16/bf16) is
# applied to the LLM decoder only. The audio embeddings are cast
# to the LLM dtype automatically when written into inputs_embeds.
# audio encoder
encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
# audio_adaptor
adaptor_out, adaptor_out_lens = self.audio_adaptor(encoder_out, encoder_out_lens)
meta_data["encoder_out"] = encoder_out
meta_data["encoder_out_lens"] = encoder_out_lens
meta_data["audio_adaptor_out"] = adaptor_out
meta_data["audio_adaptor_out_lens"] = adaptor_out_lens
input_ids = batch["input_ids"]
source_ids = batch["source_ids"]
fbank_beg = batch["fbank_beg"]
fake_token_len = batch["fake_token_len"]
if not kwargs.get("teacherforcing", False):
input_ids = source_ids
input_ids[input_ids < 0] = 0
inputs_embeds = self.llm.model.get_input_embeddings()(input_ids)
batch_size, token_num, dims = inputs_embeds.shape
fake_token_len[fake_token_len < 0] = 0
fbank_beg[fbank_beg < 0] = 0
speech_idx = 0
for batch_idx in range(batch_size):
for turn_id in range(fbank_beg.shape[1]):
fbank_beg_idx = fbank_beg[batch_idx, turn_id].item()
if fbank_beg_idx > 0:
speech_token_len = fake_token_len[batch_idx, turn_id]
speech_token = adaptor_out[speech_idx, :speech_token_len, :]
try:
inputs_embeds[
batch_idx,
fbank_beg_idx : fbank_beg_idx + speech_token_len,
:,
] = speech_token
except Exception as e:
#
logging.error(f"{str(e)}, {traceback.format_exc()}")
logging.info(
f"batch_idx: {batch_idx}, inputs_embeds: {inputs_embeds.shape}, fbank_beg_idx: {fbank_beg_idx}, speech_token_len: {speech_token_len}, adaptor_out: {adaptor_out.shape}, adaptor_out_lens: {adaptor_out_lens}, fake_token_len: {fake_token_len}, speech_lengths: {speech_lengths}"
)
speech_token_len = adaptor_out_lens[speech_idx].item()
speech_token = adaptor_out[speech_idx, :speech_token_len, :]
inputs_embeds[
batch_idx,
fbank_beg_idx : fbank_beg_idx + speech_token_len,
:,
] = speech_token
speech_idx += 1
return inputs_embeds, contents, batch, source_ids, meta_data
def get_prompt(self, hotwords: list[str], language: str = None, itn: bool = True):
"""Get prompt.
Args:
hotwords: TODO.
language: Language identifier.
itn: TODO.
"""
if len(hotwords) > 0:
hotwords = ", ".join(hotwords)
prompt = f"请结合上下文信息,更加准确地完成语音转写任务。如果没有相关信息,我们会留空。\n\n\n**上下文信息:**\n\n\n"
prompt += f"热词列表:[{hotwords}]\n"
else:
prompt = ""
if language is None:
prompt += "语音转写"
else:
prompt += f"语音转写成{language}"
if not itn:
prompt += ",不进行文本规整"
return prompt + ":"
def generate_chatml(self, prompt: str, data: Union[str, torch.Tensor]):
"""Generate chatml.
Args:
prompt: TODO.
data: TODO.
"""
if isinstance(data, str):
return [
{"role": "system", "content": "You are a helpful assistant."},
{"role": "user", "content": f"{prompt}<|startofspeech|>!{data}<|endofspeech|>"},
{"role": "assistant", "content": "null"},
]
elif isinstance(data, torch.Tensor):
return [
{"role": "system", "content": "You are a helpful assistant."},
{
"role": "user",
"content": f"{prompt}<|startofspeech|>!!<|endofspeech|>",
"audio": data,
},
{"role": "assistant", "content": "null"},
]
def inference(
self,
data_in,
data_lengths=None,
key: list = None,
tokenizer=None,
frontend=None,
**kwargs,
):
"""Run inference on input data.
Args:
data_in: Input data (audio samples, file paths, or text).
data_lengths: Lengths of each input sample in the batch.
key: Sample identifiers.
tokenizer: Tokenizer instance for text encoding/decoding.
frontend: Audio frontend for feature extraction.
**kwargs: Additional keyword arguments.
"""
prompt = self.get_prompt(
kwargs.get("hotwords", []), kwargs.get("language", None), kwargs.get("itn", True)
)
data_in = [self.generate_chatml(prompt, data) for data in data_in]
if key is None:
key = []
for _ in data_in:
chars = string.ascii_letters + string.digits
key.append("rand_key_" + "".join(random.choice(chars) for _ in range(13)))
return self.inference_llm(
data_in,
data_lengths=data_lengths,
key=key,
tokenizer=tokenizer,
frontend=frontend,
**kwargs,
)
def _inference_llm_batch(self, data_in, data_lengths, key, tokenizer, frontend, **kwargs):
"""Batched LLM decoding for multiple VAD segments at once.
Builds each segment's inputs_embeds via the single-sample
inference_prepare, left-pads them into one batch, and runs a single
llm.generate. This greatly improves GPU utilization for the small LLM
decoder (the per-segment, batch_size=1 path underuses the GPU).
CTC timestamps are not produced in batched mode.
"""
# normalize nested key (e.g. [[k1, k2, ...]]) like the single-sample path
if key is not None and len(key) > 0 and isinstance(key[0], (list, tuple)):
key = list(key[0])
embs = []
keys = []
for i, d in enumerate(data_in):
k_i = [key[i]] if key is not None and i < len(key) else None
emb_i, _c, _b, _s, _m = self.inference_prepare(
[d], data_lengths, k_i, tokenizer, frontend, **kwargs
)
embs.append(emb_i)
keys.append(key[i] if key is not None and i < len(key) else f"rand_{i}")
llm_dtype = kwargs.get("llm_dtype", "fp32")
if llm_dtype == "fp32":
llm_dtype = "fp16" if kwargs.get("fp16", False) else llm_dtype
llm_dtype = "bf16" if kwargs.get("bf16", False) else llm_dtype
dt = dtype_map[llm_dtype]
device = embs[0].device
self.llm = self.llm.to(dt)
B = len(embs)
D = embs[0].shape[-1]
Tmax = max(e.shape[1] for e in embs)
padded = torch.zeros(B, Tmax, D, device=device, dtype=dt)
attn = torch.zeros(B, Tmax, dtype=torch.long, device=device)
for i, e in enumerate(embs):
Ti = e.shape[1]
padded[i, Tmax - Ti :, :] = e[0].to(dt) # left padding
attn[i, Tmax - Ti :] = 1
autocast_device_type = resolve_autocast_device_type(kwargs.get("device", "cuda"))
with torch.autocast(
device_type=autocast_device_type,
enabled=True if llm_dtype != "fp32" else False,
dtype=dt,
):
# left padding requires explicit position_ids so each segment's real
# tokens get positions 0,1,2,... regardless of the padding length.
position_ids = attn.long().cumsum(-1) - 1
position_ids.masked_fill_(attn == 0, 1)
generated_ids = self.llm.generate(
inputs_embeds=padded,
attention_mask=attn,
position_ids=position_ids,
max_new_tokens=kwargs.get("max_length", 512),
pad_token_id=(
self.llm.config.pad_token_id
if self.llm.config.pad_token_id is not None
else self.llm.config.eos_token_id
),
**kwargs.get("llm_kwargs", {}),
)
texts = tokenizer.batch_decode(
generated_ids, skip_special_tokens=kwargs.get("skip_special_tokens", True)
)
results = []
for i, t in enumerate(texts):
t = kwargs.get("prev_text", "") + t
results.append(
{
"key": keys[i],
"text": re.sub(r"\s+", " ", t.replace("/sil", " ")),
"text_tn": re.sub(r"[^\w\s\u3000\u4e00-\u9fff]+", "", t),
}
)
return results, {}
@staticmethod
def _slice_batch_value(value, index):
if value is None:
return None
if isinstance(value, torch.Tensor) and value.ndim > 0 and value.shape[0] > index:
return value[index : index + 1]
if isinstance(value, list) and len(value) > index:
return [value[index]]
if isinstance(value, tuple) and len(value) > index:
return (value[index],)
return value
@staticmethod
def _merge_inference_meta(target, source):
for name, value in source.items():
if isinstance(value, (int, float)):
target[name] = target.get(name, 0.0) + value
elif name not in target:
target[name] = value
def _inference_llm_ctc_sequential(
self, data_in, data_lengths, key, tokenizer, frontend, **kwargs
):
"""Run multi-segment input one segment at a time when CTC timestamps are active."""
if key is not None and len(key) > 0 and isinstance(key[0], (list, tuple)):
key = list(key[0])
results = []
meta_data = {}
for i, data_i in enumerate(data_in):
key_i = [key[i]] if key is not None and i < len(key) else None
data_lengths_i = self._slice_batch_value(data_lengths, i)
results_i, meta_i = self.inference_llm(
[data_i],
data_lengths=data_lengths_i,
key=key_i,
tokenizer=tokenizer,
frontend=frontend,
**kwargs,
)
results.extend(results_i)
self._merge_inference_meta(meta_data, meta_i)
return results, meta_data
def inference_llm(
self,
data_in,
data_lengths=None,
key: list = None,
tokenizer=None,
frontend=None,
**kwargs,
):
"""Inference llm.
Args:
data_in: Input data (audio samples, file paths, or text).
data_lengths: Lengths of each input sample in the batch.
key: Sample identifiers.
tokenizer: Tokenizer instance for text encoding/decoding.
frontend: Audio frontend for feature extraction.
**kwargs: Additional keyword arguments.
"""
# Only batch when CTC timestamps are not needed; the batched path does not
# produce ctc_timestamps, so fall back to the single-sample path when a CTC
# decoder is loaded (preserves timestamp behavior).
if len(data_in) > 1:
if self.ctc_decoder is None:
return self._inference_llm_batch(
data_in, data_lengths, key, tokenizer, frontend, **kwargs
)
return self._inference_llm_ctc_sequential(
data_in, data_lengths, key, tokenizer, frontend, **kwargs
)
inputs_embeds, contents, batch, source_ids, meta_data = self.inference_prepare(
data_in, data_lengths, key, tokenizer, frontend, **kwargs
)
ctc_results = []
if self.ctc_decoder is not None:
encoder_out = meta_data["encoder_out"]
encoder_out_lens = meta_data["encoder_out_lens"]
decoder_out, decoder_out_lens = self.ctc_decoder(encoder_out, encoder_out_lens)
ctc_logits = self.ctc.log_softmax(decoder_out)
b, n, d = encoder_out.size()
if isinstance(key[0], (list, tuple)):
key = key[0]
if len(key) < b:
key = key * b
for i in range(b):
x = ctc_logits[i, : encoder_out_lens[i].item(), :]
yseq = x.argmax(dim=-1)
yseq = torch.unique_consecutive(yseq, dim=-1)
mask = yseq != self.blank_id
token_int = yseq[mask].tolist()
# Change integer-ids to tokens
text = self.ctc_tokenizer.decode(token_int)
ctc_results.append({"key": key[i], "text": text, "ctc_logits": x})
llm_dtype = kwargs.get("llm_dtype", "fp32")
if llm_dtype == "fp32":
llm_dtype = "fp16" if kwargs.get("fp16", False) else llm_dtype
llm_dtype = "bf16" if kwargs.get("bf16", False) else llm_dtype
autocast_device_type = resolve_autocast_device_type(kwargs.get("device", "cuda"))
with torch.autocast(
device_type=autocast_device_type,
enabled=True if llm_dtype != "fp32" else False,
dtype=dtype_map[llm_dtype],
):
label = contents["assistant"][-1]
self.llm = self.llm.to(dtype_map[llm_dtype])
inputs_embeds = inputs_embeds.to(dtype_map[llm_dtype])