mirror of
https://github.com/modelscope/FunASR
synced 2025-09-15 14:48:36 +08:00
474 lines
17 KiB
Python
474 lines
17 KiB
Python
import logging
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from dataclasses import dataclass
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from typing import Dict
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from typing import Iterable, Optional
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import types
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import time
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import numpy as np
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import torch
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import torch.nn.functional as F
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from torch import Tensor
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from torch import nn
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from torch.cuda.amp import autocast
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from funasr.metrics.compute_acc import compute_accuracy
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from funasr.losses.label_smoothing_loss import LabelSmoothingLoss
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from funasr.train_utils.device_funcs import force_gatherable
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from . import whisper_lib as whisper
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from funasr.utils.load_utils import load_audio_text_image_video, extract_fbank
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from funasr.register import tables
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@tables.register("model_classes", "SenseVoice")
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class SenseVoice(nn.Module):
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def __init__(self, *args, **kwargs):
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super().__init__()
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dims = kwargs.get("dims", {})
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dims = whisper.model.ModelDimensions(**dims)
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model = whisper.model.Whisper(dims=dims)
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# encoder
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model.encoder.downsample_rate = kwargs.get("downsample_rate", 4)
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model.encoder.use_padmask = kwargs.get("use_padmask", True)
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from .encoder import sense_voice_encode_forward
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model.encoder.forward = types.MethodType(sense_voice_encode_forward, model.encoder)
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# decoder
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model.decoder.use_padmask = kwargs.get("use_padmask", True)
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from .decoder import sense_voice_decode_forward
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model.decoder.forward = types.MethodType(sense_voice_decode_forward, model.decoder)
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self.model = model
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self.encoder_output_size = self.model.dims.n_audio_state
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self.activation_checkpoint = kwargs.get("activation_checkpoint", False)
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self.ignore_id = kwargs.get("ignore_id", -1)
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self.vocab_size = kwargs.get("vocab_size", -1)
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self.length_normalized_loss = kwargs.get("length_normalized_loss", True)
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self.criterion_att = LabelSmoothingLoss(
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size=self.vocab_size,
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padding_idx=self.ignore_id,
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smoothing=kwargs.get("lsm_weight", 0.0),
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normalize_length=self.length_normalized_loss,
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)
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specaug = kwargs.get("specaug", None)
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if specaug is not None:
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specaug_class = tables.specaug_classes.get(specaug)
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specaug = specaug_class(**kwargs.get("specaug_conf", {}))
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self.specaug = specaug
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def forward(
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self,
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speech: torch.Tensor,
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speech_lengths: torch.Tensor,
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text: torch.Tensor,
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text_lengths: torch.Tensor,
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**kwargs,
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):
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target_mask = kwargs.get("target_mask", None)
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# import pdb;
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# pdb.set_trace()
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if len(text_lengths.size()) > 1:
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text_lengths = text_lengths[:, 0]
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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.shape[0]
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if self.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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loss_att, acc_att, cer_att, wer_att = self._calc_att_loss(
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encoder_out, encoder_out_lens, text, text_lengths, target_mask=target_mask
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)
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loss = loss_att
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stats = {}
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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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# 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((text_lengths + 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(
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self,
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speech: torch.Tensor,
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speech_lengths: torch.Tensor,
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**kwargs,
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):
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"""Encoder. Note that this method is used by asr_inference.py
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Args:
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speech: (Batch, Length, ...)
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speech_lengths: (Batch, )
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ind: int
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"""
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with autocast(False):
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# Data augmentation
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if self.specaug is not None and self.training:
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speech, speech_lengths = self.specaug(speech, speech_lengths)
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# Forward encoder
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encoder_out, encoder_out_lens = self.model.encoder(speech.permute(0, 2, 1), speech_lengths)
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return encoder_out, encoder_out_lens
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def _calc_att_loss(
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self,
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encoder_out: torch.Tensor,
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encoder_out_lens: torch.Tensor,
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ys_pad: torch.Tensor,
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ys_pad_lens: torch.Tensor,
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**kwargs,
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):
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target_mask = kwargs.get("target_mask", None)
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stats = {}
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# 1. Forward decoder
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decoder_out = self.model.decoder(
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x=ys_pad, xa=encoder_out, hlens=encoder_out_lens, ys_in_lens=ys_pad_lens
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)
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# 2. Compute attention loss
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mask = torch.ones_like(ys_pad) * (-1)
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ys_pad_mask = (ys_pad * target_mask + mask * (1 - target_mask)).to(torch.int64)
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ys_pad_mask[ys_pad_mask == 0] = -1
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loss_att = self.criterion_att(decoder_out[:, :-1, :], ys_pad_mask[:, 1:])
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with torch.no_grad():
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preds = torch.argmax(decoder_out, -1)
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acc_att = compute_accuracy(
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preds[:, :-1], ys_pad_mask[:, 1:], ignore_label=self.ignore_id
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)
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return loss_att, acc_att, None, None
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def inference(
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self,
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data_in,
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data_lengths=None,
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key: list = None,
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tokenizer=None,
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frontend=None,
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**kwargs,
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):
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if kwargs.get("batch_size", 1) > 1:
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raise NotImplementedError("batch decoding is not implemented")
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if frontend is None and not hasattr(self, "frontend"):
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frontend_class = tables.frontend_classes.get("WhisperFrontend")
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frontend = frontend_class(
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n_mels=self.model.dims.n_mels, do_pad_trim=kwargs.get("do_pad_trim", True)
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)
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self.frontend = frontend
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else:
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frontend = frontend if frontend is not None else self.frontend
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meta_data = {}
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if (
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isinstance(data_in, torch.Tensor) and kwargs.get("data_type", "sound") == "fbank"
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): # fbank
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speech, speech_lengths = data_in, data_lengths
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if len(speech.shape) < 3:
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speech = speech[None, :, :]
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if speech_lengths is None:
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speech_lengths = speech.shape[1]
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else:
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# extract fbank feats
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time1 = time.perf_counter()
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audio_sample_list = load_audio_text_image_video(
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data_in,
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fs=frontend.fs if hasattr(frontend, "fs") else 16000,
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audio_fs=kwargs.get("fs", 16000),
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data_type=kwargs.get("data_type", "sound"),
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tokenizer=tokenizer,
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)
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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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speech, speech_lengths = extract_fbank(
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audio_sample_list, data_type=kwargs.get("data_type", "sound"), frontend=frontend
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)
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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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frame_shift = frontend.frame_shift if hasattr(frontend, "frame_shift") else 10
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lfr_n = frontend.lfr_n if hasattr(frontend, "lfr_n") else 1
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meta_data["batch_data_time"] = speech_lengths.sum().item() * frame_shift * lfr_n / 1000
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speech = speech.to(device=kwargs["device"])[0, :, :]
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speech_lengths = speech_lengths.to(device=kwargs["device"])
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DecodingOptions = kwargs.get("DecodingOptions", {})
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task = DecodingOptions.get("task", "ASR")
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if isinstance(task, str):
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task = [task]
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task = "".join([f"<|{x}|>" for x in task])
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initial_prompt = kwargs.get("initial_prompt", f"<|startoftranscript|>{task}")
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DecodingOptions["initial_prompt"] = initial_prompt
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language = DecodingOptions.get("language", None)
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language = None if language == "auto" else language
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DecodingOptions["language"] = language
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DecodingOptions["vocab_path"] = kwargs["tokenizer_conf"].get("vocab_path", None)
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if "without_timestamps" not in DecodingOptions:
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DecodingOptions["without_timestamps"] = True
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options = whisper.DecodingOptions(**DecodingOptions)
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result = whisper.decode(self.model, speech, options)
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text = f"{result.text}"
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results = []
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result_i = {"key": key[0], "text": text}
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results.append(result_i)
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return results, meta_data
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@tables.register("model_classes", "SenseVoiceRWKV")
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class SenseVoiceRWKV(nn.Module):
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def __init__(self, *args, **kwargs):
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super().__init__()
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dims = kwargs.get("dims", {})
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dims = whisper.model.ModelDimensions(**dims)
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model = whisper.model.Whisper(dims=dims)
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# encoder
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model.encoder.downsample_rate = kwargs.get("downsample_rate", 4)
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model.encoder.use_padmask = kwargs.get("use_padmask", True)
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from .encoder import sense_voice_encode_forward
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model.encoder.forward = types.MethodType(sense_voice_encode_forward, model.encoder)
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# decoder
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del model.decoder
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decoder = kwargs.get("decoder", "SenseVoiceDecoder")
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decoder_class = tables.decoder_classes.get(decoder)
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decoder = decoder_class(
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n_vocab=dims.n_vocab,
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n_ctx=dims.n_text_ctx,
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n_state=dims.n_text_state,
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n_head=dims.n_text_head,
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n_layer=dims.n_text_layer,
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**kwargs.get("decoder_conf"),
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)
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model.decoder = decoder
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self.model = model
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self.encoder_output_size = self.model.dims.n_audio_state
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self.activation_checkpoint = kwargs.get("activation_checkpoint", False)
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self.ignore_id = kwargs.get("ignore_id", -1)
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self.vocab_size = kwargs.get("vocab_size", -1)
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self.length_normalized_loss = kwargs.get("length_normalized_loss", True)
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self.criterion_att = LabelSmoothingLoss(
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size=self.vocab_size,
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padding_idx=self.ignore_id,
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smoothing=kwargs.get("lsm_weight", 0.0),
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normalize_length=self.length_normalized_loss,
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)
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specaug = kwargs.get("specaug", None)
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if specaug is not None:
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specaug_class = tables.specaug_classes.get(specaug)
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specaug = specaug_class(**kwargs.get("specaug_conf", {}))
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self.specaug = specaug
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def forward(
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self,
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speech: torch.Tensor,
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speech_lengths: torch.Tensor,
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text: torch.Tensor,
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text_lengths: torch.Tensor,
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**kwargs,
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):
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target_mask = kwargs.get("target_mask", None)
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# import pdb;
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# pdb.set_trace()
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if len(text_lengths.size()) > 1:
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text_lengths = text_lengths[:, 0]
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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, frames, _ = speech.shape
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if self.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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loss_att, acc_att, cer_att, wer_att = self._calc_att_loss(
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encoder_out, encoder_out_lens, text, text_lengths, target_mask=target_mask
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)
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loss = loss_att
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stats = {}
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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_frames"] = frames * batch_size
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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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# 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((text_lengths + 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(
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self,
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speech: torch.Tensor,
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speech_lengths: torch.Tensor,
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**kwargs,
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):
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"""Encoder. Note that this method is used by asr_inference.py
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Args:
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speech: (Batch, Length, ...)
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speech_lengths: (Batch, )
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ind: int
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"""
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with autocast(False):
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# Data augmentation
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if self.specaug is not None and self.training:
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speech, speech_lengths = self.specaug(speech, speech_lengths)
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# Forward encoder
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encoder_out, encoder_out_lens = self.model.encoder(speech.permute(0, 2, 1), speech_lengths)
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return encoder_out, encoder_out_lens
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def _calc_att_loss(
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self,
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encoder_out: torch.Tensor,
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encoder_out_lens: torch.Tensor,
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ys_pad: torch.Tensor,
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ys_pad_lens: torch.Tensor,
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**kwargs,
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):
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target_mask = kwargs.get("target_mask", None)
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stats = {}
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# 1. Forward decoder
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decoder_out = self.model.decoder(
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x=ys_pad, xa=encoder_out, hlens=encoder_out_lens, ys_in_lens=ys_pad_lens
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)
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# 2. Compute attention loss
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mask = torch.ones_like(ys_pad) * (-1)
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ys_pad_mask = (ys_pad * target_mask + mask * (1 - target_mask)).to(torch.int64)
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ys_pad_mask[ys_pad_mask == 0] = -1
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loss_att = self.criterion_att(decoder_out[:, :-1, :], ys_pad_mask[:, 1:])
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with torch.no_grad():
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preds = torch.argmax(decoder_out, -1)
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acc_att = compute_accuracy(
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preds[:, :-1], ys_pad_mask[:, 1:], ignore_label=self.ignore_id
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)
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return loss_att, acc_att, None, None
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def inference(
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self,
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data_in,
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data_lengths=None,
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key: list = None,
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tokenizer=None,
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frontend=None,
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**kwargs,
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):
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if kwargs.get("batch_size", 1) > 1:
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raise NotImplementedError("batch decoding is not implemented")
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if frontend is None and not hasattr(self, "frontend"):
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frontend_class = tables.frontend_classes.get("WhisperFrontend")
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frontend = frontend_class(
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n_mels=self.model.dims.n_mels, do_pad_trim=kwargs.get("do_pad_trim", True)
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)
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self.frontend = frontend
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else:
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frontend = frontend if frontend is not None else self.frontend
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meta_data = {}
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if (
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isinstance(data_in, torch.Tensor) and kwargs.get("data_type", "sound") == "fbank"
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): # fbank
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speech, speech_lengths = data_in, data_lengths
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if len(speech.shape) < 3:
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speech = speech[None, :, :]
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if speech_lengths is None:
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speech_lengths = speech.shape[1]
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else:
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# extract fbank feats
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time1 = time.perf_counter()
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audio_sample_list = load_audio_text_image_video(
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data_in,
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fs=frontend.fs if hasattr(frontend, "fs") else 16000,
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audio_fs=kwargs.get("fs", 16000),
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data_type=kwargs.get("data_type", "sound"),
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tokenizer=tokenizer,
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)
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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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speech, speech_lengths = extract_fbank(
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audio_sample_list, data_type=kwargs.get("data_type", "sound"), frontend=frontend
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)
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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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frame_shift = frontend.frame_shift if hasattr(frontend, "frame_shift") else 10
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lfr_n = frontend.lfr_n if hasattr(frontend, "lfr_n") else 1
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meta_data["batch_data_time"] = speech_lengths.sum().item() * frame_shift * lfr_n / 1000
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speech = speech.to(device=kwargs["device"])[0, :, :]
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speech_lengths = speech_lengths.to(device=kwargs["device"])
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DecodingOptions = kwargs.get("DecodingOptions", {})
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task = DecodingOptions.get("task", "ASR")
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if isinstance(task, str):
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task = [task]
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task = "".join([f"<|{x}|>" for x in task])
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initial_prompt = kwargs.get("initial_prompt", f"<|startoftranscript|>{task}")
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DecodingOptions["initial_prompt"] = initial_prompt
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language = DecodingOptions.get("language", None)
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language = None if language == "auto" else language
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DecodingOptions["language"] = language
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DecodingOptions["vocab_path"] = kwargs["tokenizer_conf"].get("vocab_path", None)
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if "without_timestamps" not in DecodingOptions:
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DecodingOptions["without_timestamps"] = True
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options = whisper.DecodingOptions(**DecodingOptions)
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result = whisper.decode(self.model, speech, options)
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text = f"{result.text}"
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results = []
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result_i = {"key": key[0], "text": text}
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results.append(result_i)
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return results, meta_data
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