from dataclasses import dataclass
|
from typing import Dict
|
from typing import Iterable, Optional
|
import time
|
import numpy as np
|
import torch
|
import torch.nn.functional as F
|
from torch import Tensor
|
from torch import nn
|
from . import whisper_lib as whisper
|
from funasr.utils.load_utils import load_audio_text_image_video, extract_fbank
|
|
from funasr.register import tables
|
|
|
@tables.register("model_classes", "SenseVoice")
|
class SenseVoice(nn.Module):
|
def __init__(self, *args, **kwargs):
|
super().__init__()
|
hub = kwargs.get("hub", "funasr")
|
|
dims = kwargs.get("dims", {})
|
dims = whisper.model.ModelDimensions(**dims)
|
model = whisper.model.Whisper(dims=dims)
|
|
self.model = model
|
|
self.encoder_output_size = self.model.dims.n_audio_state
|
|
def forward(self, ):
|
pass
|
|
def inference(self,
|
data_in,
|
data_lengths=None,
|
key: list = None,
|
tokenizer=None,
|
frontend=None,
|
**kwargs,
|
):
|
if kwargs.get("batch_size", 1) > 1:
|
raise NotImplementedError("batch decoding is not implemented")
|
|
if frontend is None and not hasattr(self, "frontend"):
|
frontend_class = tables.frontend_classes.get("WhisperFrontend")
|
frontend = frontend_class(n_mels=self.model.dims.n_mels, do_pad_trim=kwargs.get("do_pad_trim", True))
|
self.frontend = frontend
|
else:
|
frontend = frontend if frontend is not None else self.frontend
|
|
meta_data = {}
|
if isinstance(data_in, torch.Tensor) and kwargs.get("data_type", "sound") == "fbank": # fbank
|
speech, speech_lengths = data_in, data_lengths
|
if len(speech.shape) < 3:
|
speech = speech[None, :, :]
|
if speech_lengths is None:
|
speech_lengths = speech.shape[1]
|
else:
|
# extract fbank feats
|
time1 = time.perf_counter()
|
audio_sample_list = load_audio_text_image_video(data_in, fs=frontend.fs if hasattr(frontend, "fs") else 16000, audio_fs=kwargs.get("fs", 16000),
|
data_type=kwargs.get("data_type", "sound"),
|
tokenizer=tokenizer)
|
time2 = time.perf_counter()
|
meta_data["load_data"] = f"{time2 - time1:0.3f}"
|
speech, speech_lengths = extract_fbank(audio_sample_list, data_type=kwargs.get("data_type", "sound"),
|
frontend=frontend)
|
time3 = time.perf_counter()
|
meta_data["extract_feat"] = f"{time3 - time2:0.3f}"
|
frame_shift = frontend.frame_shift if hasattr(frontend, "frame_shift") else 10
|
lfr_n = frontend.lfr_n if hasattr(frontend, "lfr_n") else 1
|
meta_data["batch_data_time"] = speech_lengths.sum().item() * frame_shift * lfr_n / 1000
|
|
speech = speech.to(device=kwargs["device"])[0, :, :]
|
speech_lengths = speech_lengths.to(device=kwargs["device"])
|
|
task = kwargs.get("task", "ASR")
|
if isinstance(task, str):
|
task = [task]
|
task = "".join([f"<|{x}|>" for x in task])
|
initial_prompt = kwargs.get("initial_prompt", f"<|startoftranscript|>{task}")
|
language = kwargs.get("language", None)
|
language = None if language == "auto" else language
|
# if language is None:
|
# # detect the spoken language
|
# _, probs = self.model.detect_language(speech, initial_prompt=initial_prompt)
|
# print(f"Detected language: {max(probs, key=probs.get)}")
|
# language = max(probs, key=probs.get)
|
# language = language if kwargs.get("language", None) is None else kwargs.get("language")
|
|
# decode the audio
|
|
# initial_prompt = kwargs.get("initial_prompt", "<|startoftranscript|><|ASR|>")
|
|
vocab_path = kwargs.get("vocab_path", None)
|
options = whisper.DecodingOptions(language=language, fp16=False, without_timestamps=True, initial_prompt=initial_prompt, vocab_path=vocab_path)
|
|
|
result = whisper.decode(self.model, speech, options)
|
|
results = []
|
result_i = {"key": key[0], "text": result.text}
|
|
results.append(result_i)
|
|
return results, meta_data
|
|