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https://github.com/modelscope/FunASR
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v3
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funasr/models/minmo/__init__.py
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0
funasr/models/minmo/__init__.py
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funasr/models/minmo/model.py
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funasr/models/minmo/model.py
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import logging
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import os.path
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import torchaudio
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from typing import Union, Dict, List, Tuple, Optional
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import time
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import torch
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import torch.nn as nn
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import torch.nn.functional as F
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from torch.cuda.amp import autocast
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import numpy as np
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import re
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import math
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from torch.nn import CrossEntropyLoss
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from funasr.models.scama.utils import sequence_mask
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from funasr.losses.label_smoothing_loss import LabelSmoothingLoss
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from funasr.models.ctc.ctc import CTC
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from funasr.models.transformer.utils.add_sos_eos import add_sos_eos
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from funasr.metrics.compute_acc import th_accuracy, compute_accuracy
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from funasr.metrics.common import ErrorCalculator
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from funasr.train_utils.device_funcs import force_gatherable
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from funasr.utils.load_utils import load_audio_text_image_video, extract_fbank
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from funasr.utils import postprocess_utils
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from funasr.utils.datadir_writer import DatadirWriter
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from funasr.register import tables
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from funasr.train_utils.device_funcs import to_device
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from funasr.models.transformer.utils.nets_utils import make_pad_mask, pad_list
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from funasr.train_utils.set_all_random_seed import set_all_random_seed
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import traceback
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from pydub import AudioSegment
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from io import BytesIO
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try:
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import numpy as np
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from scipy.io import savemat
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except:
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pass
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dtype_map = {"bf16": torch.bfloat16, "fp16": torch.float16, "fp32": torch.float32}
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@tables.register("model_classes", "Speech2Speech")
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class Speech2Speech(nn.Module):
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""" """
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def __init__(
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self,
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audio_encoder: str = None,
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audio_encoder_conf: dict = None,
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audio_adaptor: str = None,
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audio_adaptor_conf: dict = None,
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llm: str = None,
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llm_conf: dict = None,
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input_size: int = 80,
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length_normalized_loss: bool = False,
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**kwargs,
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):
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super().__init__()
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# audio encoder
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hub = audio_encoder_conf.get("hub", None)
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self.audio_encoder_activation_checkpoint = audio_encoder_conf.get(
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"activation_checkpoint", False
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)
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if hub == "ms":
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from funasr import AutoModel
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model = AutoModel(model=audio_encoder, model_revision="master")
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# frontend = model.kwargs.get("frontend")
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audio_encoder_output_size = model.model.encoder_output_size
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audio_encoder = (
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model.model.model.encoder if hasattr(model.model, "model") else model.model.encoder
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)
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# self.frontend = frontend
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elif hub == "hf":
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pass
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else:
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encoder_class = tables.encoder_classes.get(audio_encoder)
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audio_encoder = encoder_class(input_size=input_size, **audio_encoder_conf)
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audio_encoder_output_size = audio_encoder.output_size()
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freeze = audio_encoder_conf.get("freeze", True)
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freeze_layer_num = int(audio_encoder_conf.get("freeze_layer_num", -1))
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# if freeze_layer_num > 0:
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# freeze_layer_num = range(freeze_layer_num)
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if freeze:
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for name, param in audio_encoder.named_parameters():
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if freeze_layer_num > 0:
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idx = re.search(r"\.\d+\.", name)
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if idx is not None:
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beg, end = idx.regs[0]
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layer_id = int(name[beg + 1 : end - 1])
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if layer_id < freeze_layer_num:
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param.requires_grad = False
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elif "ln_post." not in name:
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param.requires_grad = False
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else:
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param.requires_grad = False
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audio_encoder.eval()
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self.audio_encoder = audio_encoder
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# llm
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self.llm = None
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from transformers import AutoModelForCausalLM, AutoTokenizer, AutoConfig
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init_param_path = llm_conf.get("init_param_path", "vicuna-7b-v1.5")
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llm_load_kwargs = llm_conf.get("load_kwargs", {})
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if not llm_conf.get("low_cpu", False):
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model = AutoModelForCausalLM.from_pretrained(
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init_param_path,
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load_in_8bit=None,
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device_map=None,
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use_cache=None,
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**llm_load_kwargs,
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)
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else:
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import os
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if int(os.environ.get("RANK", 0)) == 0:
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model = AutoModelForCausalLM.from_pretrained(
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init_param_path,
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load_in_8bit=None,
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device_map="cpu",
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use_cache=None,
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**llm_load_kwargs,
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)
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else:
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llm_config = AutoConfig.from_pretrained(init_param_path)
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model = AutoModelForCausalLM.from_config(llm_config)
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freeze = llm_conf.get("freeze", True)
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if freeze:
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for name, param in model.named_parameters():
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param.requires_grad = False
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model.eval()
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logging.info(f"use_lora: {llm_conf.get('use_lora', False)}")
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if llm_conf.get("use_lora", False):
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from omegaconf import OmegaConf, DictConfig
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lora_conf = llm_conf.get("lora_conf", {})
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if isinstance(lora_conf, (OmegaConf, DictConfig)):
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lora_conf = OmegaConf.to_container(lora_conf, resolve=True)
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from peft import get_peft_model, LoraConfig, TaskType, PeftConfig, PeftModel
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lora_init_param_path = lora_conf.get("init_param_path", None)
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if lora_init_param_path is not None:
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logging.info(f"lora_init_param_path: {lora_init_param_path}")
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model = PeftModel.from_pretrained(model, lora_init_param_path)
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for name, param in model.named_parameters():
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if not lora_conf.get("freeze_lora", False):
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if "lora_" in name:
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param.requires_grad = True
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else:
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peft_config = LoraConfig(**lora_conf)
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model = get_peft_model(model, peft_config)
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model.print_trainable_parameters()
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if llm_conf.get("activation_checkpoint", False):
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model.gradient_checkpointing_enable()
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self.llm_dtype = llm_conf.get("llm_dtype", "fp32")
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self.llm = model.to(dtype_map[self.llm_dtype])
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llm_dim = model.get_input_embeddings().weight.shape[-1]
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# adaptor
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adaptor_class = tables.adaptor_classes.get(audio_adaptor)
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audio_adaptor_conf["encoder_dim"] = audio_encoder_output_size
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audio_adaptor_conf["llm_dim"] = llm_dim
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audio_adaptor = adaptor_class(**audio_adaptor_conf)
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init_param_path = audio_adaptor_conf.get("init_param_path", None)
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if init_param_path is not None:
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src_state = torch.load(init_param_path, map_location="cpu")
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flag = audio_adaptor.load_state_dict(src_state, strict=False)
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logging.info(f"Loading audio_adaptor ckpt: {init_param_path}, status: {flag}")
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freeze = audio_adaptor_conf.get("freeze", False)
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if freeze:
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for name, param in audio_adaptor.named_parameters():
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param.requires_grad = False
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audio_adaptor.eval()
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self.audio_adaptor = audio_adaptor
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self.error_calculator = None
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self.length_normalized_loss = length_normalized_loss
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self.beam_search = None
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import os
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rank = int(os.environ.get("RANK", 0))
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logging.info(f"rank: {rank}, model is builded.")
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def forward(
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self,
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speech: torch.Tensor = None,
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speech_lengths: torch.Tensor = None,
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input_ids: torch.Tensor = None,
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attention_mask: torch.Tensor = None,
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labels_ids: torch.Tensor = None,
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fbank_beg: torch.Tensor = None,
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fbank_mask: torch.Tensor = None,
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**kwargs,
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):
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"""Encoder + Decoder + Calc loss
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Args:
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speech: (Batch, Length, ...)
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speech_lengths: (Batch, )
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text: (Batch, Length)
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text_lengths: (Batch,)
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"""
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# import pdb
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#
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# pdb.set_trace()
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batch_size, token_num = input_ids.shape
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stats = {}
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input_ids[input_ids < 0] = 0
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inputs_embeds = self.llm.model.get_input_embeddings()(input_ids)
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if speech is not None:
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if len(speech_lengths.size()) > 1:
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speech_lengths = speech_lengths[:, 0]
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batch_size_speech, frames, _ = speech.shape
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# audio encoder
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if self.audio_encoder_activation_checkpoint:
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from torch.utils.checkpoint import checkpoint
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encoder_out, encoder_out_lens = checkpoint(
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self.encode, speech, speech_lengths, use_reentrant=False
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)
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else:
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encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
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# audio_adaptor
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encoder_out, encoder_out_lens = self.audio_adaptor(encoder_out, encoder_out_lens)
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batch_size, token_num, dims = inputs_embeds.shape
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fake_token_len = kwargs.get("fake_token_len")
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fake_token_len[fake_token_len < 0] = 0
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fbank_beg[fbank_beg < 0] = 0
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speech_idx = 0
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for batch_idx in range(batch_size):
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for turn_id in range(fbank_beg.shape[1]):
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fbank_beg_idx = fbank_beg[batch_idx, turn_id].item()
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if fbank_beg_idx > 0:
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speech_token_len = fake_token_len[batch_idx, turn_id]
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speech_token = encoder_out[speech_idx, :speech_token_len, :]
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try:
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inputs_embeds[
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batch_idx, fbank_beg_idx : fbank_beg_idx + speech_token_len, :
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] = speech_token
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except Exception as e:
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#
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logging.error(f"{str(e)}, {traceback.format_exc()}")
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logging.info(
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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}"
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)
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# import pdb;
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# pdb.set_trace()
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speech_token_len = encoder_out_lens[speech_idx].item()
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speech_token = encoder_out[speech_idx, :speech_token_len, :]
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inputs_embeds[
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batch_idx, fbank_beg_idx : fbank_beg_idx + speech_token_len, :
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] = speech_token
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speech_idx += 1
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stats["batch_size_speech"] = batch_size_speech
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stats["batch_size_x_frames"] = frames * batch_size_speech
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stats["batch_size_real_frames"] = speech_lengths.sum().item()
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stats["padding_frames"] = stats["batch_size_x_frames"] - stats["batch_size_real_frames"]
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with torch.cuda.amp.autocast(
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enabled=True if self.llm_dtype != "fp32" else False, dtype=dtype_map[self.llm_dtype]
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):
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labels_ids[labels_ids == -1] = -100
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attention_mask[attention_mask < 0] = 0
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model_outputs = self.llm(
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inputs_embeds=inputs_embeds.to(dtype_map[self.llm_dtype]),
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attention_mask=attention_mask,
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labels=labels_ids,
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)
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loss = model_outputs.loss
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with torch.no_grad():
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preds = torch.argmax(model_outputs.logits, -1)
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acc_att = compute_accuracy(preds[:, :-1], labels_ids[:, 1:], ignore_label=-100)
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stats["acc"] = acc_att
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stats["loss"] = torch.clone(loss.detach())
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stats["batch_size"] = batch_size
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stats["batch_size_x_tokens"] = token_num * batch_size
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stats["batch_size_real_tokens"] = attention_mask.sum().item()
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stats["padding_tokens"] = stats["batch_size_x_tokens"] - stats["batch_size_real_tokens"]
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dialog_turns = (fbank_beg > 0).sum(-1)
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dialog_turns_max = torch.max(dialog_turns).int().item()
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dialog_turns_avg = dialog_turns.sum().item() / batch_size
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stats["dialog_turns_max"] = dialog_turns_max
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stats["dialog_turns_avg"] = dialog_turns_avg
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# force_gatherable: to-device and to-tensor if scalar for DataParallel
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if self.length_normalized_loss:
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batch_size = int((labels_ids > 0 + 1).sum())
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loss, stats, weight = force_gatherable((loss, stats, batch_size), loss.device)
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return loss, stats, weight
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def encode(self, speech, speech_lengths):
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# audio encoder
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encoder_out, encoder_out_lens = self.audio_encoder(speech.permute(0, 2, 1), speech_lengths)
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return encoder_out, encoder_out_lens
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def data_template(self, data):
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system, user, assistant = [], [], []
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for i, item in enumerate(data):
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role = item["role"]
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content = item["content"]
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if role == "system":
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system.append(content)
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elif role == "user":
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if "audio" in item:
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audio = item["audio"]
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content = [content, audio]
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user.append(content)
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elif role == "assistant":
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assistant.append(content)
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system = system * len(user)
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contents = {
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"system": system,
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"user": user,
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"assistant": assistant,
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}
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return contents
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def data_load_speech(self, contents: dict, tokenizer, frontend, meta_data={}, **kwargs):
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system = contents["system"]
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user = contents["user"]
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assistant = contents["assistant"]
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pattern = re.compile(r"(<\|startofspeech\|>.*?<\|endofspeech\|>)")
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input_ids, labels, fbank, fbank_lens, fbank_mask, fbank_beg, fake_token_len = (
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[],
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[],
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[],
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[],
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[],
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[],
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[],
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)
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input_source_ids = []
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for i, (system_prompt, user_prompt, target_out) in enumerate(zip(system, user, assistant)):
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if i >= kwargs.get("multiturn_num_max", 5):
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break
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if len(input_ids) > kwargs.get("max_token_length", 1500):
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break
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if isinstance(user_prompt, (list, tuple)):
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user_prompt, audio = user_prompt
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if i == 0:
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if kwargs.get("infer_with_assistant_input", False):
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source_input = f"<|im_start|>system\n{system_prompt}<|im_end|>\n<|im_start|>user\n{user_prompt}"
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else:
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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"
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else:
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if kwargs.get("infer_with_assistant_input", False):
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source_input = f"<|im_start|>user\n{user_prompt}"
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else:
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source_input = (
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f"<|im_start|>user\n{user_prompt}<|im_end|>\n<|im_start|>assistant\n"
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)
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splits = pattern.split(source_input)
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source_ids = []
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fbank_i = []
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fbank_mask_i = []
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fake_token_len_i = 0
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fbank_beg_i = -1
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fbank_lens_i = []
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speech, speech_lengths = [], []
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for k, sub_str in enumerate(splits):
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if not sub_str.startswith("<|startofspeech|>"):
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sub_token = tokenizer.encode(sub_str)
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source_ids += sub_token
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fbank_mask_i += [0] * len(sub_token)
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else:
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sub_str = sub_str.replace("<|startofspeech|>", "").replace(
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"<|endofspeech|>", ""
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)
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if sub_str.startswith("!"):
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sub_str = sub_str[1:]
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if sub_str.startswith("!"): # !!: audio sample point
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sub_str = audio
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try:
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time1 = time.perf_counter()
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data_src = load_audio_text_image_video(sub_str, fs=frontend.fs)
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time2 = time.perf_counter()
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meta_data["load_data"] = f"{time2 - time1:0.3f}"
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except Exception as e:
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logging.error(f"Loading wav failed! {str(e)}, {traceback.format_exc()}")
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speech, speech_lengths = extract_fbank(
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data_src,
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data_type=kwargs.get("data_type", "sound"),
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frontend=frontend,
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is_final=True,
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) # speech: [b, T, d]
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time3 = time.perf_counter()
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meta_data["extract_feat"] = f"{time3 - time2:0.3f}"
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meta_data["batch_data_time"] = (
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speech_lengths.sum().item()
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* frontend.frame_shift
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* frontend.lfr_n
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/ 1000
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)
|
||||
|
||||
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:
|
||||
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]
|
||||
# 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)
|
||||
|
||||
meta_data["audio_adaptor_out"] = encoder_out
|
||||
meta_data["audio_adaptor_out_lens"] = 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])
|
||||
llm_kwargs = kwargs.get("llm_kwargs", {})
|
||||
if not kwargs.get("tearchforing", False):
|
||||
|
||||
generated_ids = self.llm.generate(
|
||||
inputs_embeds=inputs_embeds,
|
||||
max_new_tokens=kwargs.get("max_length", 512),
|
||||
**llm_kwargs,
|
||||
)
|
||||
# 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,
|
||||
**llm_kwargs,
|
||||
)
|
||||
|
||||
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(r"[^\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)
|
||||
|
||||
|
||||
class LayerNorm(nn.LayerNorm):
|
||||
def __init__(self, *args, **kwargs):
|
||||
super().__init__(*args, **kwargs)
|
||||
|
||||
def forward(self, input):
|
||||
output = F.layer_norm(
|
||||
input.float(),
|
||||
self.normalized_shape,
|
||||
self.weight.float() if self.weight is not None else None,
|
||||
self.bias.float() if self.bias is not None else None,
|
||||
self.eps,
|
||||
)
|
||||
return output.type_as(input)
|
||||
Loading…
Reference in New Issue
Block a user