From 5de8bfdcd8a617ac13c13478505401bbf4e57472 Mon Sep 17 00:00:00 2001
From: 游雁 <zhifu.gzf@alibaba-inc.com>
Date: 星期四, 13 六月 2024 15:38:17 +0800
Subject: [PATCH] decoding
---
funasr/models/llm_asr/model.py | 777 ++++++++++++++++++++++++++++++++++++++++++++++++-----------
1 files changed, 630 insertions(+), 147 deletions(-)
diff --git a/funasr/models/llm_asr/model.py b/funasr/models/llm_asr/model.py
index a903262..15969e3 100644
--- a/funasr/models/llm_asr/model.py
+++ b/funasr/models/llm_asr/model.py
@@ -6,32 +6,38 @@
import torch.nn as nn
import torch.nn.functional as F
from torch.cuda.amp import autocast
-
+import re
from funasr.models.scama.utils import sequence_mask
from funasr.losses.label_smoothing_loss import LabelSmoothingLoss
from funasr.models.ctc.ctc import CTC
from funasr.models.transformer.utils.add_sos_eos import add_sos_eos
from funasr.metrics.compute_acc import th_accuracy, compute_accuracy
-# from funasr.models.e2e_asr_common import ErrorCalculator
+from funasr.metrics.common import ErrorCalculator
from funasr.train_utils.device_funcs import force_gatherable
from funasr.utils.load_utils import load_audio_text_image_video, extract_fbank
from funasr.utils import postprocess_utils
from funasr.utils.datadir_writer import DatadirWriter
from funasr.register import tables
+from funasr.train_utils.device_funcs import to_device
+import traceback
+
+dtype_map = {"bf16": torch.bfloat16, "fp16": torch.float16, "fp32": torch.float32}
-@tables.register("model_classes", "LLMASRNAR")
-class LLMASRNAR(nn.Module):
+@tables.register("model_classes", "LLMASR")
+class LLMASR(nn.Module):
""" """
-
+
def __init__(
self,
specaug: str = None,
specaug_conf: dict = None,
normalize: str = None,
normalize_conf: dict = None,
- encoder: str = None,
- encoder_conf: dict = None,
+ audio_encoder: str = None,
+ audio_encoder_conf: dict = None,
+ audio_adaptor: str = None,
+ audio_adaptor_conf: dict = None,
decoder: str = None,
decoder_conf: dict = None,
ctc: str = None,
@@ -39,8 +45,6 @@
ctc_weight: float = 0.5,
llm: str = None,
llm_conf: dict = None,
- adaptor: str = None,
- adaptor_conf: dict = None,
input_size: int = 80,
vocab_size: int = -1,
ignore_id: int = -1,
@@ -59,34 +63,42 @@
# postencoder: Optional[AbsPostEncoder] = None,
**kwargs,
):
-
+
super().__init__()
-
+
if specaug is not None:
specaug_class = tables.specaug_classes.get(specaug)
specaug = specaug_class(**specaug_conf)
if normalize is not None:
normalize_class = tables.normalize_classes.get(normalize)
normalize = normalize_class(**normalize_conf)
-
+
# audio encoder
- hub = encoder_conf.get("hub", None)
- if hub == "funasr":
+ hub = audio_encoder_conf.get("hub", None)
+ if hub == "ms":
from funasr import AutoModel
- init_param_path = encoder_conf.get("hub", "iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch")
- model = AutoModel(model=init_param_path, model_revision="v2.0.4")
+
+ model = AutoModel(model=audio_encoder, model_revision="master")
# frontend = model.kwargs.get("frontend")
- model.model.decoder = None
-
- self.audio_encoder = model.model
+ audio_encoder_output_size = model.model.encoder_output_size
+
+ audio_encoder = model.model.model.encoder
+
# self.frontend = frontend
-
+
elif hub == "hf":
pass
else:
- encoder_class = tables.encoder_classes.get(encoder)
- encoder = encoder_class(input_size=input_size, **encoder_conf)
- encoder_output_size = encoder.output_size()
+ 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 name, param in audio_encoder.named_parameters():
+ param.requires_grad = False
+ audio_encoder.eval()
+
+ self.audio_encoder = audio_encoder
# llm
hub = llm_conf.get("hub", "hf")
@@ -95,6 +107,7 @@
from transformers import AutoModelForCausalLM, AutoTokenizer, AutoConfig
init_param_path = llm_conf.get("init_param_path", "vicuna-7b-v1.5")
+
model = AutoModelForCausalLM.from_pretrained(
init_param_path,
load_in_8bit=None,
@@ -107,14 +120,14 @@
param.requires_grad = False
model.eval()
self.llm = model
-
+
# adaptor
- adaptor_class = tables.adaptor_classes.get(adaptor)
- adaptor = adaptor_class(**adaptor_conf)
-
- self.adaptor = adaptor
-
-
+ adaptor_class = tables.adaptor_classes.get(audio_adaptor)
+ audio_adaptor_conf["encoder_dim"] = audio_encoder_output_size
+ audio_adaptor = adaptor_class(**audio_adaptor_conf)
+
+ self.audio_adaptor = audio_adaptor
+
self.blank_id = blank_id
self.sos = sos if sos is not None else vocab_size - 1
self.eos = eos if eos is not None else vocab_size - 1
@@ -122,8 +135,6 @@
self.ignore_id = ignore_id
self.specaug = specaug
self.normalize = normalize
- self.encoder = encoder
-
self.criterion_att = LabelSmoothingLoss(
size=vocab_size,
@@ -131,17 +142,12 @@
smoothing=lsm_weight,
normalize_length=length_normalized_loss,
)
- #
- # if report_cer or report_wer:
- # self.error_calculator = ErrorCalculator(
- # token_list, sym_space, sym_blank, report_cer, report_wer
- # )
- #
+
self.error_calculator = None
self.length_normalized_loss = length_normalized_loss
self.beam_search = None
-
+
def forward(
self,
speech: torch.Tensor,
@@ -149,7 +155,7 @@
text: torch.Tensor,
text_lengths: torch.Tensor,
input_ids: torch.Tensor,
- attention_mask:torch.Tensor,
+ attention_mask: torch.Tensor,
labels_ids: torch.Tensor,
label_mask: torch.Tensor,
audio_mask: torch.Tensor,
@@ -168,41 +174,43 @@
text_lengths = text_lengths[:, 0]
if len(speech_lengths.size()) > 1:
speech_lengths = speech_lengths[:, 0]
-
+
batch_size = speech.shape[0]
-
+
# audio encoder
- encoder_out, encoder_out_lens = self.encode(speech, speech_lengths, audio_mask=audio_mask)
-
- # adaptor
- encoder_out = self.adaptor(encoder_out)
+ encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
- if input_ids is not None:
- input_ids[input_ids == -1] = 0
- if hasattr(self.llm.model, "embed_tokens"):
- inputs_embeds = self.llm.model.embed_tokens(input_ids)
- elif hasattr(self.llm.model.model, "embed_tokens"):
- inputs_embeds = self.llm.model.model.embed_tokens(input_ids)
- else:
- inputs_embeds = self.llm.model.model.model.embed_tokens(input_ids)
+ # audio_adaptor
+ encoder_out = self.audio_adaptor(encoder_out)
- if audio_mask is not None:
- batch_size, token_num, dims = inputs_embeds.shape
- _, l, _ = encoder_out.shape
- encoder_outs_pad = F.pad(encoder_out, (0, 0, token_num-l-1, 1, 0, 0), value=0.0)
- inputs_embeds = encoder_outs_pad * audio_mask[:, :, None] + inputs_embeds * (~audio_mask[:, :, None])
- inputs_embeds = F.pad(inputs_embeds[:, 1:, :], (0, 0, 0, 1, 0, 0), value=0.0)
+ input_ids[input_ids == -1] = 0
+ input_ids[input_ids == -100] = 0
+ if hasattr(self.llm.model, "embed_tokens"):
+ inputs_embeds = self.llm.model.embed_tokens(input_ids)
+ elif hasattr(self.llm.model.model, "embed_tokens"):
+ inputs_embeds = self.llm.model.model.embed_tokens(input_ids)
+ else:
+ inputs_embeds = self.llm.model.model.model.embed_tokens(input_ids)
- model_outputs = self.llm(inputs_embeds=inputs_embeds, attention_mask=attention_mask, labels=labels_ids)
+ if audio_mask is not None:
+ batch_size, token_num, dims = inputs_embeds.shape
+ _, l, _ = encoder_out.shape
+ # [audio, bos, prompt, input, pad]
+ encoder_outs_pad = F.pad(encoder_out, (0, 0, 0, token_num - l, 0, 0), value=0.0)
+ inputs_embeds = encoder_outs_pad * audio_mask[:, :, None] + inputs_embeds * (
+ 1.0 - audio_mask[:, :, None]
+ )
+
+ model_outputs = self.llm(
+ inputs_embeds=inputs_embeds, attention_mask=attention_mask, labels=labels_ids
+ )
loss = model_outputs.loss
-
stats = {}
- if self.metric:
- 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
+ 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())
@@ -211,45 +219,40 @@
batch_size = int((text_lengths + 1).sum())
loss, stats, weight = force_gatherable((loss, stats, batch_size), loss.device)
return loss, stats, weight
-
+
def encode(
- self, speech: torch.Tensor, speech_lengths: torch.Tensor, **kwargs,
- ) -> Tuple[torch.Tensor, torch.Tensor]:
-
- audio_mask = kwargs.get("audio_mask")
- audio_token_lengths = audio_mask.sum(-1)
+ self,
+ speech: torch.Tensor,
+ speech_lengths: torch.Tensor,
+ **kwargs,
+ ):
+ speech = speech.permute(0, 2, 1)
+ res = self.audio_encoder(speech)
+ if isinstance(res, (list, tuple)):
+ encoder_out, encoder_out_lens = res[0], res[1]
+ else:
+ encoder_out, encoder_out_lens = res, speech_lengths
+ return encoder_out, encoder_out_lens
- batch = {"speech": speech, "speech_lengths": speech_lengths}
- enc, enc_lens = self.audio_encoder.encode(**batch)
- enc_mask = sequence_mask(enc_lens, enc.size(1), device=enc.device)[:, None, :]
- pre_acoustic_embeds, pre_token_length, _, _ = self.audio_encoder.predictor(enc,
- mask=enc_mask,
- target_label_length=audio_token_lengths,
- )
+ def inference(
+ self,
+ data_in,
+ data_lengths=None,
+ key: list = None,
+ tokenizer=None,
+ frontend=None,
+ **kwargs,
+ ):
- return pre_acoustic_embeds, pre_token_length
+ prompt = kwargs.get("prompt", "Transcribe speech to text.")
-
- def inference(self,
- data_in,
- data_lengths=None,
- key: list = None,
- tokenizer=None,
- frontend=None,
- **kwargs,
- ):
-
if kwargs.get("batch_size", 1) > 1:
raise NotImplementedError("batch decoding is not implemented")
-
- # init beamsearch
- if self.beam_search is None:
- logging.info("enable beam_search")
- self.init_beam_search(**kwargs)
- self.nbest = kwargs.get("nbest", 1)
-
+
meta_data = {}
- if isinstance(data_in, torch.Tensor) and kwargs.get("data_type", "sound") == "fbank": # fbank
+ if (
+ isinstance(data_in, torch.Tensor) and kwargs.get("data_type", "sound") == "fbank"
+ ): # fbank
speech, speech_lengths = data_in, data_lengths
if len(speech.shape) < 3:
speech = speech[None, :, :]
@@ -258,63 +261,543 @@
else:
# extract fbank feats
time1 = time.perf_counter()
- audio_sample_list = load_audio_text_image_video(data_in, fs=frontend.fs, audio_fs=kwargs.get("fs", 16000),
- data_type=kwargs.get("data_type", "sound"),
- tokenizer=tokenizer)
+ audio_sample_list = load_audio_text_image_video(
+ data_in,
+ fs=frontend.fs,
+ audio_fs=kwargs.get("fs", 16000),
+ data_type=kwargs.get("data_type", "sound"),
+ tokenizer=tokenizer,
+ )
time2 = time.perf_counter()
meta_data["load_data"] = f"{time2 - time1:0.3f}"
- speech, speech_lengths = extract_fbank(audio_sample_list, data_type=kwargs.get("data_type", "sound"),
- frontend=frontend)
+ speech, speech_lengths = extract_fbank(
+ audio_sample_list, data_type=kwargs.get("data_type", "sound"), frontend=frontend
+ )
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
-
+ meta_data["batch_data_time"] = (
+ speech_lengths.sum().item() * frontend.frame_shift * frontend.lfr_n / 1000
+ )
+
speech = speech.to(device=kwargs["device"])
speech_lengths = speech_lengths.to(device=kwargs["device"])
+
# Encoder
encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
- if isinstance(encoder_out, tuple):
- encoder_out = encoder_out[0]
-
- # c. Passed the encoder result and the beam search
- nbest_hyps = self.beam_search(
- x=encoder_out[0], maxlenratio=kwargs.get("maxlenratio", 0.0), minlenratio=kwargs.get("minlenratio", 0.0)
+
+ # adaptor
+ encoder_out = self.audio_adaptor(encoder_out)
+
+ prompt_pre = "USER: \nINSTRUCTION: {}\nINPUT: ".format(prompt)
+ prompt_ids = tokenizer.encode(prompt_pre)
+ prompt_length = len(prompt_ids)
+ prompt_ids = torch.tensor(prompt_ids, dtype=torch.int64).to(kwargs["device"])
+
+ if hasattr(self.llm.model, "embed_tokens"):
+ inputs_embeds = self.llm.model.embed_tokens(prompt_ids)
+ elif hasattr(self.llm.model.model, "embed_tokens"):
+ inputs_embeds = self.llm.model.model.embed_tokens(prompt_ids)
+ else:
+ inputs_embeds = self.llm.model.model.model.embed_tokens(prompt_ids)
+
+ inputs_embeds = torch.cat(
+ (inputs_embeds[None, :, :], encoder_out), dim=1
+ ) # [prompt, audio]
+ attention_mask = torch.ones(inputs_embeds.size()[:-1], dtype=torch.long).to(
+ kwargs["device"]
)
-
- nbest_hyps = nbest_hyps[: self.nbest]
-
+
+ preds = self.llm.generate(
+ inputs_embeds=inputs_embeds,
+ max_length=kwargs.get("max_length", 200),
+ max_new_tokens=kwargs.get("max_new_tokens", 200),
+ num_beams=kwargs.get("num_beams", 4),
+ do_sample=kwargs.get("do_sample", False),
+ min_length=kwargs.get("min_length", 1),
+ top_p=kwargs.get("top_p", 1.0),
+ repetition_penalty=kwargs.get("repetition_penalty", 1.0),
+ length_penalty=kwargs.get("length_penalty", 1.0),
+ temperature=kwargs.get("temperature", 1.0),
+ attention_mask=attention_mask,
+ bos_token_id=tokenizer.bos_token_id,
+ eos_token_id=tokenizer.eos_token_id,
+ pad_token_id=tokenizer.pad_token_id,
+ )
+
+ text = tokenizer.batch_decode(preds, add_special_tokens=False, skip_special_tokens=True)
+
+ text = text[0].split(": ")[-1]
+ text = text.strip()
+
+ # preds = torch.argmax(model_outputs.logits, -1)
+
+ ibest_writer = None
+ if kwargs.get("output_dir") is not None:
+ if not hasattr(self, "writer"):
+ self.writer = DatadirWriter(kwargs.get("output_dir"))
+ ibest_writer = self.writer[f"{0 + 1}best_recog"]
+
results = []
- b, n, d = encoder_out.size()
- for i in range(b):
-
- for nbest_idx, hyp in enumerate(nbest_hyps):
- ibest_writer = None
- if kwargs.get("output_dir") is not None:
- if not hasattr(self, "writer"):
- self.writer = DatadirWriter(kwargs.get("output_dir"))
- ibest_writer = self.writer[f"{nbest_idx + 1}best_recog"]
-
- # remove sos/eos and get results
- last_pos = -1
- if isinstance(hyp.yseq, list):
- token_int = hyp.yseq[1:last_pos]
- else:
- token_int = hyp.yseq[1:last_pos].tolist()
-
- # remove blank symbol id, which is assumed to be 0
- token_int = list(filter(lambda x: x != self.eos and x != self.sos and x != self.blank_id, token_int))
-
- # Change integer-ids to tokens
- token = tokenizer.ids2tokens(token_int)
- text = tokenizer.tokens2text(token)
-
- text_postprocessed, _ = postprocess_utils.sentence_postprocess(token)
- result_i = {"key": key[i], "token": token, "text": text_postprocessed}
- results.append(result_i)
-
- if ibest_writer is not None:
- ibest_writer["token"][key[i]] = " ".join(token)
- ibest_writer["text"][key[i]] = text_postprocessed
-
+ result_i = {"key": key[0], "text": text}
+ results.append(result_i)
+
+ if ibest_writer is not None:
+ ibest_writer["text"][key[0]] = text
+
return results, meta_data
+
+@tables.register("model_classes", "LLMASR2")
+class LLMASR2(nn.Module):
+ """ """
+
+ def __init__(
+ self,
+ specaug: str = None,
+ specaug_conf: dict = None,
+ normalize: str = None,
+ normalize_conf: dict = None,
+ audio_encoder: str = None,
+ audio_encoder_conf: dict = None,
+ audio_adaptor: str = None,
+ audio_adaptor_conf: dict = None,
+ decoder: str = None,
+ decoder_conf: dict = None,
+ ctc: str = None,
+ ctc_conf: dict = None,
+ ctc_weight: float = 0.5,
+ llm: str = None,
+ llm_conf: dict = None,
+ input_size: int = 80,
+ vocab_size: int = -1,
+ ignore_id: int = -1,
+ blank_id: int = 0,
+ sos: int = 1,
+ eos: int = 2,
+ lsm_weight: float = 0.0,
+ length_normalized_loss: bool = False,
+ report_cer: bool = True,
+ report_wer: bool = True,
+ sym_space: str = "<space>",
+ sym_blank: str = "<blank>",
+ # extract_feats_in_collect_stats: bool = True,
+ share_embedding: bool = False,
+ # preencoder: Optional[AbsPreEncoder] = None,
+ # postencoder: Optional[AbsPostEncoder] = None,
+ **kwargs,
+ ):
+
+ super().__init__()
+
+ # audio encoder
+ hub = audio_encoder_conf.get("hub", None)
+ if hub == "ms":
+ from funasr import AutoModel
+
+ model = AutoModel(model=audio_encoder, model_revision="master")
+ # frontend = model.kwargs.get("frontend")
+ audio_encoder_output_size = model.model.encoder_output_size
+
+ audio_encoder = model.model.model.encoder
+
+ # self.frontend = frontend
+
+ elif hub == "hf":
+ pass
+ 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)
+ freeze_layer_num = int(audio_encoder_conf.get("freeze_layer_num", -1))
+ if freeze_layer_num > 0:
+ freeze_layer_num = range(freeze_layer_num)
+
+ if freeze:
+ for name, param in audio_encoder.named_parameters():
+ if isinstance(freeze_layer_num, (list, tuple)):
+ idx = re.search(r"\.\d+\.", name)
+ if idx is not None:
+ beg, end = idx.regs[0]
+ layer_id = int(name[beg + 1 : end - 1])
+ if layer_id in freeze_layer_num:
+ param.requires_grad = False
+ else:
+ param.requires_grad = False
+ else:
+ param.requires_grad = False
+
+ audio_encoder.eval()
+
+ self.audio_encoder = audio_encoder
+
+ # llm
+ self.llm = None
+
+ from transformers import AutoModelForCausalLM, AutoTokenizer, AutoConfig
+
+ init_param_path = llm_conf.get("init_param_path", "vicuna-7b-v1.5")
+
+ model = AutoModelForCausalLM.from_pretrained(
+ init_param_path,
+ load_in_8bit=None,
+ device_map=None,
+ use_cache=None,
+ )
+ freeze = llm_conf.get("freeze", True)
+ if freeze:
+ for name, param in model.named_parameters():
+ param.requires_grad = False
+ model.eval()
+ self.llm = model
+ llm_dim = model.get_input_embeddings().weight.shape[-1]
+ self.llm_dtype = llm_conf.get("llm_dtype", "fp32")
+
+ # adaptor
+ adaptor_class = tables.adaptor_classes.get(audio_adaptor)
+ audio_adaptor_conf["encoder_dim"] = audio_encoder_output_size
+ audio_adaptor_conf["llm_dim"] = llm_dim
+ audio_adaptor = adaptor_class(**audio_adaptor_conf)
+ init_param_path = audio_adaptor_conf.get("init_param_path", None)
+ if init_param_path is not None:
+ src_state = torch.load(init_param_path, map_location="cpu")
+ flag = audio_adaptor.load_state_dict(src_state, strict=False)
+ logging.info(f"Loading audio_adaptor ckpt: {init_param_path}, status: {flag}")
+
+ self.audio_adaptor = audio_adaptor
+
+ self.error_calculator = None
+
+ self.length_normalized_loss = length_normalized_loss
+ self.beam_search = None
+
+ def forward(
+ self,
+ speech: torch.Tensor,
+ speech_lengths: torch.Tensor,
+ input_ids: torch.Tensor,
+ attention_mask: torch.Tensor,
+ labels_ids: torch.Tensor,
+ fbank_beg: torch.Tensor,
+ fbank_mask: torch.Tensor,
+ **kwargs,
+ ) -> Tuple[torch.Tensor, Dict[str, torch.Tensor], torch.Tensor]:
+ """Encoder + Decoder + Calc loss
+ Args:
+ speech: (Batch, Length, ...)
+ speech_lengths: (Batch, )
+ text: (Batch, Length)
+ text_lengths: (Batch,)
+ """
+ # import pdb;
+ # pdb.set_trace()
+ if len(speech_lengths.size()) > 1:
+ speech_lengths = speech_lengths[:, 0]
+
+ batch_size, frames, _ = speech.shape
+
+ # audio encoder
+ encoder_out, encoder_out_lens = self.audio_encoder(speech.permute(0, 2, 1), speech_lengths)
+
+ # audio_adaptor
+ encoder_out, encoder_out_lens = self.audio_adaptor(encoder_out, encoder_out_lens)
+
+ input_ids[input_ids < 0] = 0
+ inputs_embeds = self.llm.model.get_input_embeddings()(input_ids)
+
+ batch_size, token_num, dims = inputs_embeds.shape
+ fbank_mask[fbank_mask < 0] = 0
+ fbank_fake_lens = fbank_mask.sum(-1).to(torch.int32)
+ # _, l, _ = encoder_out.shape
+ for batch_idx in range(batch_size):
+
+ fbank_fake_len = fbank_fake_lens[batch_idx].item()
+ fbank_beg_idx = fbank_beg[batch_idx, 0].item()
+ min_len = min(fbank_fake_len, inputs_embeds.shape[1] - fbank_beg_idx)
+
+ try:
+ inputs_embeds[batch_idx, fbank_beg_idx : fbank_beg_idx + min_len, :] = encoder_out[
+ batch_idx, :min_len, :
+ ]
+ 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}, min_len: {min_len}, fbank_fake_len: {fbank_fake_len}, encoder_out: {encoder_out.shape}, encoder_out_lens: {encoder_out_lens[batch_idx].item()}"
+ )
+ fbank_fake_len = encoder_out_lens[batch_idx].item()
+ min_len = min(fbank_fake_len, min_len)
+ inputs_embeds[batch_idx, fbank_beg_idx : fbank_beg_idx + min_len, :] = encoder_out[
+ batch_idx, :min_len, :
+ ]
+
+ with torch.cuda.amp.autocast(
+ 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, attention_mask=attention_mask, labels=labels_ids
+ )
+ loss = model_outputs.loss
+
+ stats = {}
+ 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_frames"] = frames * batch_size
+ stats["batch_size_real_frames"] = speech_lengths.sum().item()
+ stats["padding_frames"] = stats["batch_size_x_frames"] - stats["batch_size_real_frames"]
+ 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"]
+
+ # 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 data_template(self, data):
+ system, user, assistant = [], [], []
+ for i, item in enumerate(data):
+ role = item["role"]
+ content = item["content"]
+ if role == "system":
+ system.append(content)
+ elif role == "user":
+ 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):
+
+ system = contents["system"]
+ user = contents["user"]
+ assistant = contents["assistant"]
+ pattern = re.compile(r"(<\|startofspeech\|>.*?<\|endofspeech\|>)")
+ input_ids, labels, source_ids, target_ids, fbank, fbank_lens, fbank_mask, fbank_beg = (
+ [],
+ [],
+ [],
+ [],
+ [],
+ [],
+ [],
+ [],
+ )
+
+ for i, (system_prompt, user_prompt, target_out) in enumerate(zip(system, user, assistant)):
+
+ 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"
+
+ splits = pattern.split(source_input)
+ source_ids_i = []
+ fbank_mask_i = []
+ fbank_beg_i = []
+ fbank_lens_i = []
+ # target_ids_i = []
+ for k, sub_str in enumerate(splits):
+ if not sub_str.startswith("<|startofspeech|>"):
+ sub_token = tokenizer.encode(sub_str)
+ source_ids_i += sub_token
+ fbank_mask_i += [0] * len(sub_token)
+ else:
+ sub_str = sub_str.replace("<|startofspeech|>", "").replace(
+ "<|endofspeech|>", ""
+ )
+ if sub_str.startswith("!"):
+ try:
+ time1 = time.perf_counter()
+ data_src = load_audio_text_image_video(sub_str[1:], fs=frontend.fs)
+ 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 kwargs.get("permute", True):
+ speech = speech.permute(0, 2, 1)
+
+ olens = 1 + (speech_lengths[0].item() - 3 + 2 * 1) // 2
+ olens = 1 + (olens - 3 + 2 * 1) // 2
+ sub_token_len = (olens - 1) // 2 + 1
+ sub_token = [0] * sub_token_len
+ fbank_beg_i = [len(source_ids_i)]
+ source_ids_i += sub_token
+ fbank_mask_i += [1] * len(sub_token)
+
+ source_mask = [-100] * len(source_ids_i)
+ target_out = f"{target_out}<|im_end|>"
+ target_ids = tokenizer.encode(target_out)
+ input_ids += source_ids_i + target_ids
+ labels += source_mask + target_ids
+ fbank_mask += fbank_mask_i
+ fbank_beg.append(fbank_beg_i)
+
+ 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]
+ source_ids = torch.tensor(source_ids_i, dtype=torch.int64)
+ target_ids = torch.tensor(target_ids, dtype=torch.int64)
+
+ fbank = speech[0, :, :]
+ fbank_lens = speech_lengths
+ fbank_mask = torch.tensor(fbank_mask, dtype=torch.float32)
+ fbank_beg = torch.tensor(fbank_beg, dtype=torch.int32)
+
+ output = {
+ "speech": fbank[None, :, :],
+ "speech_lengths": fbank_lens[:, None],
+ "fbank_mask": fbank_mask[None, :],
+ "fbank_beg": fbank_beg[None,],
+ "input_ids": input_ids[None, :],
+ "attention_mask": attention_mask[None, :],
+ "labels_ids": labels[None, :],
+ "source_ids": source_ids[None, :],
+ "target_ids": target_ids[None, :],
+ }
+
+ return output
+
+ def inference(
+ self,
+ data_in,
+ data_lengths=None,
+ key: list = None,
+ tokenizer=None,
+ frontend=None,
+ **kwargs,
+ ):
+
+ meta_data = {}
+ prompt = kwargs.get("prompt", None)
+
+ if kwargs.get("batch_size", 1) > 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"]
+ speech_lengths = batch["speech_lengths"][:, 0]
+ # fp16
+ if kwargs.get("fp16", False):
+ speech = speech.to(torch.float16)
+ elif kwargs.get("bf16", False):
+ speech = speech.to(torch.bfloat16)
+ encoder_out, encoder_out_lens = self.audio_encoder(speech.permute(0, 2, 1), speech_lengths)
+
+ # audio_adaptor
+ encoder_out, encoder_out_lens = self.audio_adaptor(encoder_out, encoder_out_lens)
+
+ input_ids = batch["input_ids"]
+ source_ids = batch["source_ids"]
+ if not kwargs.get("tearchforing", 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
+ fbank_beg = batch["fbank_beg"]
+ for batch_idx in range(batch_size):
+
+ min_len = encoder_out_lens[batch_idx].item()
+ fbank_beg_idx = fbank_beg[batch_idx]
+ inputs_embeds[batch_idx, fbank_beg_idx : fbank_beg_idx + min_len, :] = encoder_out[
+ batch_idx, :min_len, :
+ ]
+
+ 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
+
+ with torch.cuda.amp.autocast(
+ enabled=True if llm_dtype != "fp32" else False, dtype=dtype_map[llm_dtype]
+ ):
+ label = contents["assistant"][0]
+ self.llm = self.llm.to(dtype_map[llm_dtype])
+ inputs_embeds = inputs_embeds.to(dtype_map[llm_dtype])
+
+ if not kwargs.get("tearchforing", False):
+
+ generated_ids = self.llm.generate(
+ inputs_embeds=inputs_embeds, max_new_tokens=kwargs.get("max_length", 512)
+ )
+ # generated_ids = [
+ # output_ids[len(input_id) :]
+ # for input_id, output_ids in zip(input_ids, generated_ids)
+ # ]
+ response = tokenizer.batch_decode(
+ generated_ids, skip_special_tokens=kwargs.get("skip_special_tokens", True)
+ )[0]
+
+ loss = None
+ else:
+
+ labels_ids = batch["labels_ids"]
+ labels_ids[labels_ids == -1] = -100
+ attention_mask = batch.get("attention_mask", None)
+ # attention_mask = attention_mask.to(dtype_map[llm_dtype])
+ model_outputs = self.llm(
+ inputs_embeds=inputs_embeds, attention_mask=attention_mask, labels=labels_ids
+ )
+
+ preds = torch.argmax(model_outputs.logits, -1)[:, source_ids.shape[1] :]
+ response = tokenizer.batch_decode(
+ preds,
+ add_special_tokens=False,
+ skip_special_tokens=kwargs.get("skip_special_tokens", True),
+ )[0]
+ loss = model_outputs.loss.item()
+
+ ibest_writer = None
+ if kwargs.get("output_dir") is not None:
+ if not hasattr(self, "writer"):
+ self.writer = DatadirWriter(kwargs.get("output_dir"))
+ ibest_writer = self.writer[f"{0 + 1}best_recog"]
+
+ results = []
+ response_clean = re.sub("[^\w\s\u3000\u4e00-\u9fff]+", "", response)
+ result_i = {"key": key[0], "text": response, "text_tn": response_clean, "label": label}
+ if loss is not None:
+ result_i["loss"] = loss
+ results.append(result_i)
+
+ if ibest_writer is not None:
+ ibest_writer["text"][key[0]] = response
+ ibest_writer["label"][key[0]] = label
+ ibest_writer["text_tn"][key[0]] = response_clean
+
+ return results, meta_data
--
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