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游雁 2024-07-08 17:37:07 +08:00
parent 259ea7523f
commit 037270ea44

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@ -1409,602 +1409,12 @@ class LLMASR4(nn.Module):
return results, meta_data
# @tables.register("model_classes", "LLMASR5")
# class LLMASR5(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,
# 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 if hasattr(model.model, "model") else 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 freeze_layer_num > 0:
# 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 < freeze_layer_num:
# param.requires_grad = False
# elif "ln_post." not in name:
# 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_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]
#
# # 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
#
# self.eos = kwargs.get("eos", 151645)
#
# 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_speech, frames, _ = speech.shape
# batch_size, token_num = input_ids.shape
#
# with torch.cuda.amp.autocast(enabled=False):
# # audio encoder
# 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)
#
# 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 = 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}"
# )
# # import pdb;
# # pdb.set_trace()
# 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
#
# 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.to(dtype_map[self.llm_dtype]),
# attention_mask=attention_mask,
# labels=labels_ids,
# )
# loss = model_outputs.loss
#
# codec = kwargs.get("codec")
# codec_len = kwargs.get("codec_len")
# if len(codec_len.size()) > 1:
# codec_len = codec_len[:, 0]
# hidden_states = model_outputs.hidden_states[-1].float()
#
# target_ids = []
# target_ids_len = []
# hidden_states_select = []
# for batch_idx in range(labels_ids.shape[0]):
# beg_i = 0
# end_i = 0
# for token_idx in range(labels_ids.shape[1]):
# token_int = labels_ids[batch_idx, token_idx].item()
# if token_int == self.eos:
# target_ids_i = labels_ids[batch_idx, beg_i:end_i]
# target_ids_len_i = end_i - beg_i
# target_ids_len.append(target_ids_len_i)
# target_ids.append(target_ids_i)
# hidden_states_i = hidden_states[batch_idx, beg_i - 1 : end_i - 1, :]
# hidden_states_select.append(hidden_states_i)
# beg_i = end_i
# continue
#
# end_i += 1
# if token_int <= 0:
# beg_i += 1
#
# target_ids = torch.nn.utils.rnn.pad_sequence(
# target_ids, batch_first=True, padding_value=-100
# )
# hidden_states_select = torch.nn.utils.rnn.pad_sequence(
# hidden_states_select, batch_first=True, padding_value=0.0
# )
#
# 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_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"]
# 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 encode(self, speech, speech_lengths):
# # audio encoder
# encoder_out, encoder_out_lens = self.audio_encoder(speech.permute(0, 2, 1), speech_lengths)
#
# return encoder_out, encoder_out_lens
#
# 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, 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 i == 0:
# 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"
# else:
# source_input = f"<|im_start|>user\n{user_prompt}<|im_end|>\n<|im_start|>assistant\n"
#
# splits = pattern.split(source_input)
# source_ids = []
# fbank_i = []
# fbank_mask_i = []
# fake_token_len_i = 0
# fbank_beg_i = -1
# fbank_lens_i = []
# 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("!"): # !!bytes
# sub_str = eval(sub_str[1:])
# try:
# time1 = time.perf_counter()
# data_src = load_audio_text_image_video(sub_str, 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)
# if speech_lengths > kwargs.get("max_source_length", 5500):
# # logging.info(
# # f"speech_lengths > max_source_length: {speech_lengths}>{self.max_source_length}, {item}"
# # )
# badcase_flag = True
#
# 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
# 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 = speech[0, :, :]
# # fbank_lens = torch.tensor(fbank_lens, dtype=torch.int32)
# 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,
# ):
#
# 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"]
# if len(speech) > 0:
# 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)
# # audio encoder
# 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)
#
# 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("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
#
# 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}"
# )
# # import pdb;
# # pdb.set_trace()
# 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
# return inputs_embeds, contents, batch, source_ids, meta_data
#
# def inference(
# self,
# data_in,
# data_lengths=None,
# key: list = None,
# tokenizer=None,
# frontend=None,
# **kwargs,
# ):
#
# inputs_embeds, contents, batch, source_ids, meta_data = self.inference_prepare(
# data_in, data_lengths, key, tokenizer, frontend, **kwargs
# )
#
# 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"][-1]
# 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.replace("\n", " ")
# ibest_writer["label"][key[0]] = label.replace("\n", " ")
# ibest_writer["text_tn"][key[0]] = response_clean
#
# return results, meta_data
class Swish(torch.nn.Module):
"""Construct an Swish object."""
def forward(self, x):
"""Return Swich activation function."""
return x * torch.sigmoid(x)
@tables.register("model_classes", "LLMASR5")
@ -2118,6 +1528,13 @@ class LLMASR5(nn.Module):
# audio decoder related
self.concat_emb_hidden = audio_decoder_conf.get("concat_emb_hidden", False)
self.concat_emb_hidden_norm = audio_decoder_conf.get("concat_emb_hidden_norm", False)
if self.concat_emb_hidden_norm:
self.hidden_norm = torch.nn.LayerNorm(llm_dim)
self.fusion_dropout = nn.Dropout(audio_decoder_conf.get("fusion_drop_rate", 0.0))
self.emb_norm = torch.nn.LayerNorm(self._output_size)
self.fusion_norm = torch.nn.LayerNorm(self._output_size)
self.fusion_act = Swish()
self.codebook_dim = audio_decoder_conf.get("codebook_dim", 1024)
self.codebook_size = audio_decoder_conf.get("codebook_size", 4096)
self.lm_out_voc_size = self.codebook_size + 1
@ -2299,14 +1716,15 @@ class LLMASR5(nn.Module):
# import pdb
#
# pdb.set_trace()
stats = {}
input_ids[input_ids < 0] = 0
inputs_embeds = self.llm.model.get_input_embeddings()(input_ids)
batch_size, token_num, dims = inputs_embeds.shape
if speech is not None:
if len(speech_lengths.size()) > 1:
speech_lengths = speech_lengths[:, 0]
batch_size_speech, frames, _ = speech.shape
batch_size, token_num = input_ids.shape
with torch.cuda.amp.autocast(enabled=False):
# audio encoder
@ -2315,7 +1733,6 @@ class LLMASR5(nn.Module):
# 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
@ -2349,6 +1766,11 @@ class LLMASR5(nn.Module):
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"]
with torch.cuda.amp.autocast(
enabled=True if self.llm_dtype != "fp32" else False, dtype=dtype_map[self.llm_dtype]
):
@ -2403,11 +1825,19 @@ class LLMASR5(nn.Module):
target_emb = self.llm.model.get_input_embeddings()(target_ids)
hidden_states_select = hidden_states_select.to(device=input_ids.device)
if self.concat_emb_hidden:
hidden_states_select = torch.concat((hidden_states_select, target_emb), dim=-1)
if not self.concat_emb_hidden_norm:
hidden_states_select = torch.concat((hidden_states_select, target_emb), dim=-1)
else:
outs = self.hidden_norm(hidden_states_select)
outs = self.fusion_dropout(self.fusion_act(outs))
# emb = model_outputs.hidden_states[0]
emb = self.fusion_dropout(self.fusion_act(self.emb_norm(target_emb)))
outs = self.audio_decoder_in_proj(torch.cat([outs, emb], dim=-1))
hidden_states_select = self.fusion_act(self.fusion_norm(outs))
nll, logits, target, target_lengths = self.nll(
hidden_states_select, target_ids_len, codec[:, :, None], codec_len
)
output_mask = (
~make_pad_mask(target_lengths, maxlen=target_lengths.max())
.to(hidden_states_select.device)
@ -2417,7 +1847,6 @@ class LLMASR5(nn.Module):
denom = total if self.length_normalized_loss else batch_size
loss = (nll * output_mask).sum() / denom
stats = {}
with torch.no_grad():
preds = torch.argmax(model_outputs.logits, -1)
acc_att = compute_accuracy(preds[:, :-1], labels_ids[:, 1:], ignore_label=-100)
@ -2432,10 +1861,6 @@ class LLMASR5(nn.Module):
stats["loss"] = torch.clone(loss.detach())
stats["batch_size"] = batch_size
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"]
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"]