update paraformer streaming code

This commit is contained in:
haoneng.lhn 2023-04-27 00:21:20 +08:00
parent b78d47f1ef
commit 7584bbd6f3
5 changed files with 196 additions and 363 deletions

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@ -19,7 +19,6 @@ from typing import List
import numpy as np
import torch
import torchaudio
from typeguard import check_argument_types
from funasr.fileio.datadir_writer import DatadirWriter
@ -40,11 +39,12 @@ from funasr.utils.types import str2bool
from funasr.utils.types import str2triple_str
from funasr.utils.types import str_or_none
from funasr.utils import asr_utils, wav_utils, postprocess_utils
from funasr.models.frontend.wav_frontend import WavFrontend
from funasr.models.e2e_asr_paraformer import BiCifParaformer, ContextualParaformer
from funasr.models.frontend.wav_frontend import WavFrontend, WavFrontendOnline
from funasr.export.models.e2e_asr_paraformer import Paraformer as Paraformer_export
np.set_printoptions(threshold=np.inf)
class Speech2Text:
"""Speech2Text class
@ -89,7 +89,7 @@ class Speech2Text:
)
frontend = None
if asr_train_args.frontend is not None and asr_train_args.frontend_conf is not None:
frontend = WavFrontend(cmvn_file=cmvn_file, **asr_train_args.frontend_conf)
frontend = WavFrontendOnline(cmvn_file=cmvn_file, **asr_train_args.frontend_conf)
logging.info("asr_model: {}".format(asr_model))
logging.info("asr_train_args: {}".format(asr_train_args))
@ -189,8 +189,7 @@ class Speech2Text:
@torch.no_grad()
def __call__(
self, cache: dict, speech: Union[torch.Tensor, np.ndarray], speech_lengths: Union[torch.Tensor, np.ndarray] = None,
begin_time: int = 0, end_time: int = None,
self, cache: dict, speech: Union[torch.Tensor], speech_lengths: Union[torch.Tensor] = None
):
"""Inference
@ -201,38 +200,57 @@ class Speech2Text:
"""
assert check_argument_types()
# Input as audio signal
if isinstance(speech, np.ndarray):
speech = torch.tensor(speech)
if self.frontend is not None:
feats, feats_len = self.frontend.forward(speech, speech_lengths)
feats = to_device(feats, device=self.device)
feats_len = feats_len.int()
self.asr_model.frontend = None
results = []
cache_en = cache["encoder"]
if speech.shape[1] < 16 * 60 and cache["is_final"]:
cache["last_chunk"] = True
feats = cache["feats"]
feats_len = torch.tensor([feats.shape[1]])
else:
feats = speech
feats_len = speech_lengths
lfr_factor = max(1, (feats.size()[-1] // 80) - 1)
feats_len = cache["encoder"]["stride"] + cache["encoder"]["pad_left"] + cache["encoder"]["pad_right"]
feats = feats[:,cache["encoder"]["start_idx"]:cache["encoder"]["start_idx"]+feats_len,:]
feats_len = torch.tensor([feats_len])
batch = {"speech": feats, "speech_lengths": feats_len, "cache": cache}
if self.frontend is not None:
feats, feats_len = self.frontend.forward(speech, speech_lengths, cache_en["is_final"])
feats = to_device(feats, device=self.device)
feats_len = feats_len.int()
self.asr_model.frontend = None
else:
feats = speech
feats_len = speech_lengths
# a. To device
if feats.shape[1] != 0:
if cache_en["is_final"]:
if feats.shape[1] + cache_en["chunk_size"][2] < cache_en["chunk_size"][1]:
cache_en["last_chunk"] = True
else:
# first chunk
feats_chunk1 = feats[:, :cache_en["chunk_size"][1], :]
feats_len = torch.tensor([feats_chunk1.shape[1]])
results_chunk1 = self.infer(feats_chunk1, feats_len, cache)
# last chunk
cache_en["last_chunk"] = True
feats_chunk2 = feats[:, -(feats.shape[1] + cache_en["chunk_size"][2] - cache_en["chunk_size"][1]):, :]
feats_len = torch.tensor([feats_chunk2.shape[1]])
results_chunk2 = self.infer(feats_chunk2, feats_len, cache)
return results_chunk1 + results_chunk2
results = self.infer(feats, feats_len, cache)
return results
@torch.no_grad()
def infer(self, feats: Union[torch.Tensor], feats_len: Union[torch.Tensor], cache: List = None):
batch = {"speech": feats, "speech_lengths": feats_len}
batch = to_device(batch, device=self.device)
# b. Forward Encoder
enc, enc_len = self.asr_model.encode_chunk(feats, feats_len, cache)
enc, enc_len = self.asr_model.encode_chunk(feats, feats_len, cache=cache)
if isinstance(enc, tuple):
enc = enc[0]
# assert len(enc) == 1, len(enc)
enc_len_batch_total = torch.sum(enc_len).item() * self.encoder_downsampling_factor
predictor_outs = self.asr_model.calc_predictor_chunk(enc, cache)
pre_acoustic_embeds, pre_token_length, alphas, pre_peak_index = predictor_outs[0], predictor_outs[1], \
predictor_outs[2], predictor_outs[3]
pre_token_length = pre_token_length.floor().long()
pre_acoustic_embeds, pre_token_length= predictor_outs[0], predictor_outs[1]
if torch.max(pre_token_length) < 1:
return []
decoder_outs = self.asr_model.cal_decoder_with_predictor_chunk(enc, pre_acoustic_embeds, cache)
@ -279,166 +297,12 @@ class Speech2Text:
text = self.tokenizer.tokens2text(token)
else:
text = None
results.append((text, token, token_int, hyp, enc_len_batch_total, lfr_factor))
results.append(text)
# assert check_return_type(results)
return results
class Speech2TextExport:
"""Speech2TextExport class
"""
def __init__(
self,
asr_train_config: Union[Path, str] = None,
asr_model_file: Union[Path, str] = None,
cmvn_file: Union[Path, str] = None,
lm_train_config: Union[Path, str] = None,
lm_file: Union[Path, str] = None,
token_type: str = None,
bpemodel: str = None,
device: str = "cpu",
maxlenratio: float = 0.0,
minlenratio: float = 0.0,
dtype: str = "float32",
beam_size: int = 20,
ctc_weight: float = 0.5,
lm_weight: float = 1.0,
ngram_weight: float = 0.9,
penalty: float = 0.0,
nbest: int = 1,
frontend_conf: dict = None,
hotword_list_or_file: str = None,
**kwargs,
):
# 1. Build ASR model
asr_model, asr_train_args = ASRTask.build_model_from_file(
asr_train_config, asr_model_file, cmvn_file, device
)
frontend = None
if asr_train_args.frontend is not None and asr_train_args.frontend_conf is not None:
frontend = WavFrontend(cmvn_file=cmvn_file, **asr_train_args.frontend_conf)
logging.info("asr_model: {}".format(asr_model))
logging.info("asr_train_args: {}".format(asr_train_args))
asr_model.to(dtype=getattr(torch, dtype)).eval()
token_list = asr_model.token_list
logging.info(f"Decoding device={device}, dtype={dtype}")
# 5. [Optional] Build Text converter: e.g. bpe-sym -> Text
if token_type is None:
token_type = asr_train_args.token_type
if bpemodel is None:
bpemodel = asr_train_args.bpemodel
if token_type is None:
tokenizer = None
elif token_type == "bpe":
if bpemodel is not None:
tokenizer = build_tokenizer(token_type=token_type, bpemodel=bpemodel)
else:
tokenizer = None
else:
tokenizer = build_tokenizer(token_type=token_type)
converter = TokenIDConverter(token_list=token_list)
logging.info(f"Text tokenizer: {tokenizer}")
# self.asr_model = asr_model
self.asr_train_args = asr_train_args
self.converter = converter
self.tokenizer = tokenizer
self.device = device
self.dtype = dtype
self.nbest = nbest
self.frontend = frontend
model = Paraformer_export(asr_model, onnx=False)
self.asr_model = model
@torch.no_grad()
def __call__(
self, speech: Union[torch.Tensor, np.ndarray], speech_lengths: Union[torch.Tensor, np.ndarray] = None
):
"""Inference
Args:
speech: Input speech data
Returns:
text, token, token_int, hyp
"""
assert check_argument_types()
# Input as audio signal
if isinstance(speech, np.ndarray):
speech = torch.tensor(speech)
if self.frontend is not None:
feats, feats_len = self.frontend.forward(speech, speech_lengths)
feats = to_device(feats, device=self.device)
feats_len = feats_len.int()
self.asr_model.frontend = None
else:
feats = speech
feats_len = speech_lengths
enc_len_batch_total = feats_len.sum()
lfr_factor = max(1, (feats.size()[-1] // 80) - 1)
batch = {"speech": feats, "speech_lengths": feats_len}
# a. To device
batch = to_device(batch, device=self.device)
decoder_outs = self.asr_model(**batch)
decoder_out, ys_pad_lens = decoder_outs[0], decoder_outs[1]
results = []
b, n, d = decoder_out.size()
for i in range(b):
am_scores = decoder_out[i, :ys_pad_lens[i], :]
yseq = am_scores.argmax(dim=-1)
score = am_scores.max(dim=-1)[0]
score = torch.sum(score, dim=-1)
# pad with mask tokens to ensure compatibility with sos/eos tokens
yseq = torch.tensor(
yseq.tolist(), device=yseq.device
)
nbest_hyps = [Hypothesis(yseq=yseq, score=score)]
for hyp in nbest_hyps:
assert isinstance(hyp, (Hypothesis)), type(hyp)
# remove sos/eos and get results
last_pos = -1
if isinstance(hyp.yseq, list):
token_int = hyp.yseq[1:last_pos]
else:
token_int = hyp.yseq[1:last_pos].tolist()
# remove blank symbol id, which is assumed to be 0
token_int = list(filter(lambda x: x != 0 and x != 2, token_int))
# Change integer-ids to tokens
token = self.converter.ids2tokens(token_int)
if self.tokenizer is not None:
text = self.tokenizer.tokens2text(token)
else:
text = None
results.append((text, token, token_int, hyp, enc_len_batch_total, lfr_factor))
return results
def inference(
maxlenratio: float,
minlenratio: float,
@ -536,8 +400,6 @@ def inference_modelscope(
**kwargs,
):
assert check_argument_types()
ncpu = kwargs.get("ncpu", 1)
torch.set_num_threads(ncpu)
if word_lm_train_config is not None:
raise NotImplementedError("Word LM is not implemented")
@ -580,11 +442,9 @@ def inference_modelscope(
penalty=penalty,
nbest=nbest,
)
if export_mode:
speech2text = Speech2TextExport(**speech2text_kwargs)
else:
speech2text = Speech2Text(**speech2text_kwargs)
speech2text = Speech2Text(**speech2text_kwargs)
def _load_bytes(input):
middle_data = np.frombuffer(input, dtype=np.int16)
middle_data = np.asarray(middle_data)
@ -599,7 +459,33 @@ def inference_modelscope(
offset = i.min + abs_max
array = np.frombuffer((middle_data.astype(dtype) - offset) / abs_max, dtype=np.float32)
return array
def _prepare_cache(cache: dict = {}, chunk_size=[5,10,5], batch_size=1):
if len(cache) > 0:
return cache
cache_en = {"start_idx": 0, "cif_hidden": torch.zeros((batch_size, 1, 320)),
"cif_alphas": torch.zeros((batch_size, 1)), "chunk_size": chunk_size, "last_chunk": False,
"feats": torch.zeros((batch_size, chunk_size[0] + chunk_size[2], 560))}
cache["encoder"] = cache_en
cache_de = {"decode_fsmn": None}
cache["decoder"] = cache_de
return cache
def _cache_reset(cache: dict = {}, chunk_size=[5,10,5], batch_size=1):
if len(cache) > 0:
cache_en = {"start_idx": 0, "cif_hidden": torch.zeros((batch_size, 1, 320)),
"cif_alphas": torch.zeros((batch_size, 1)), "chunk_size": chunk_size, "last_chunk": False,
"feats": torch.zeros((batch_size, chunk_size[0] + chunk_size[2], 560))}
cache["encoder"] = cache_en
cache_de = {"decode_fsmn": None}
cache["decoder"] = cache_de
return cache
def _forward(
data_path_and_name_and_type,
raw_inputs: Union[np.ndarray, torch.Tensor] = None,
@ -610,123 +496,35 @@ def inference_modelscope(
):
# 3. Build data-iterator
if data_path_and_name_and_type is not None and data_path_and_name_and_type[2] == "bytes":
raw_inputs = _load_bytes(data_path_and_name_and_type[0])
raw_inputs = torch.tensor(raw_inputs)
if data_path_and_name_and_type is None and raw_inputs is not None:
if isinstance(raw_inputs, np.ndarray):
raw_inputs = torch.tensor(raw_inputs)
is_final = False
cache = {}
chunk_size = [5, 10, 5]
if param_dict is not None and "cache" in param_dict:
cache = param_dict["cache"]
if param_dict is not None and "is_final" in param_dict:
is_final = param_dict["is_final"]
if param_dict is not None and "chunk_size" in param_dict:
chunk_size = param_dict["chunk_size"]
if data_path_and_name_and_type is not None and data_path_and_name_and_type[2] == "bytes":
raw_inputs = _load_bytes(data_path_and_name_and_type[0])
raw_inputs = torch.tensor(raw_inputs)
if data_path_and_name_and_type is not None and data_path_and_name_and_type[2] == "sound":
raw_inputs = torchaudio.load(data_path_and_name_and_type[0])[0][0]
is_final = True
if data_path_and_name_and_type is None and raw_inputs is not None:
if isinstance(raw_inputs, np.ndarray):
raw_inputs = torch.tensor(raw_inputs)
# 7 .Start for-loop
# FIXME(kamo): The output format should be discussed about
raw_inputs = torch.unsqueeze(raw_inputs, axis=0)
input_lens = torch.tensor([raw_inputs.shape[1]])
asr_result_list = []
results = []
asr_result = ""
wait = True
if len(cache) == 0:
cache["encoder"] = {"start_idx": 0, "pad_left": 0, "stride": 10, "pad_right": 5, "cif_hidden": None, "cif_alphas": None, "is_final": is_final, "left": 0, "right": 0}
cache_de = {"decode_fsmn": None}
cache["decoder"] = cache_de
cache["first_chunk"] = True
cache["speech"] = []
cache["accum_speech"] = 0
if raw_inputs is not None:
if len(cache["speech"]) == 0:
cache["speech"] = raw_inputs
else:
cache["speech"] = torch.cat([cache["speech"], raw_inputs], dim=0)
cache["accum_speech"] += len(raw_inputs)
while cache["accum_speech"] >= 960:
if cache["first_chunk"]:
if cache["accum_speech"] >= 14400:
speech = torch.unsqueeze(cache["speech"], axis=0)
speech_length = torch.tensor([len(cache["speech"])])
cache["encoder"]["pad_left"] = 5
cache["encoder"]["pad_right"] = 5
cache["encoder"]["stride"] = 10
cache["encoder"]["left"] = 5
cache["encoder"]["right"] = 0
results = speech2text(cache, speech, speech_length)
cache["accum_speech"] -= 4800
cache["first_chunk"] = False
cache["encoder"]["start_idx"] = -5
cache["encoder"]["is_final"] = False
wait = False
else:
if is_final:
cache["encoder"]["stride"] = len(cache["speech"]) // 960
cache["encoder"]["pad_left"] = 0
cache["encoder"]["pad_right"] = 0
speech = torch.unsqueeze(cache["speech"], axis=0)
speech_length = torch.tensor([len(cache["speech"])])
results = speech2text(cache, speech, speech_length)
cache["accum_speech"] = 0
wait = False
else:
break
else:
if cache["accum_speech"] >= 19200:
cache["encoder"]["start_idx"] += 10
cache["encoder"]["stride"] = 10
cache["encoder"]["pad_left"] = 5
cache["encoder"]["pad_right"] = 5
cache["encoder"]["left"] = 0
cache["encoder"]["right"] = 0
speech = torch.unsqueeze(cache["speech"], axis=0)
speech_length = torch.tensor([len(cache["speech"])])
results = speech2text(cache, speech, speech_length)
cache["accum_speech"] -= 9600
wait = False
else:
if is_final:
cache["encoder"]["is_final"] = True
if cache["accum_speech"] >= 14400:
cache["encoder"]["start_idx"] += 10
cache["encoder"]["stride"] = 10
cache["encoder"]["pad_left"] = 5
cache["encoder"]["pad_right"] = 5
cache["encoder"]["left"] = 0
cache["encoder"]["right"] = cache["accum_speech"] // 960 - 15
speech = torch.unsqueeze(cache["speech"], axis=0)
speech_length = torch.tensor([len(cache["speech"])])
results = speech2text(cache, speech, speech_length)
cache["accum_speech"] -= 9600
wait = False
else:
cache["encoder"]["start_idx"] += 10
cache["encoder"]["stride"] = cache["accum_speech"] // 960 - 5
cache["encoder"]["pad_left"] = 5
cache["encoder"]["pad_right"] = 0
cache["encoder"]["left"] = 0
cache["encoder"]["right"] = 0
speech = torch.unsqueeze(cache["speech"], axis=0)
speech_length = torch.tensor([len(cache["speech"])])
results = speech2text(cache, speech, speech_length)
cache["accum_speech"] = 0
wait = False
else:
break
if len(results) >= 1:
asr_result += results[0][0]
if asr_result == "":
asr_result = "sil"
if wait:
asr_result = "waiting_for_more_voice"
item = {'key': "utt", 'value': asr_result}
asr_result_list.append(item)
else:
return []
cache = _prepare_cache(cache, chunk_size=chunk_size, batch_size=1)
cache["encoder"]["is_final"] = is_final
asr_result = speech2text(cache, raw_inputs, input_lens)
item = {'key': "utt", 'value': asr_result}
asr_result_list.append(item)
if is_final:
cache = _cache_reset(cache, chunk_size=chunk_size, batch_size=1)
return asr_result_list
return _forward
@ -921,4 +719,3 @@ if __name__ == "__main__":
# rec_result = inference_16k_pipline(audio_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav')
# print(rec_result)

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@ -712,9 +712,9 @@ class ParaformerOnline(Paraformer):
def calc_predictor_chunk(self, encoder_out, cache=None):
pre_acoustic_embeds, pre_token_length, alphas, pre_peak_index = \
pre_acoustic_embeds, pre_token_length = \
self.predictor.forward_chunk(encoder_out, cache["encoder"])
return pre_acoustic_embeds, pre_token_length, alphas, pre_peak_index
return pre_acoustic_embeds, pre_token_length
def cal_decoder_with_predictor_chunk(self, encoder_out, sematic_embeds, cache=None):
decoder_outs = self.decoder.forward_chunk(

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@ -6,9 +6,11 @@ from typing import Union
import logging
import torch
import torch.nn as nn
import torch.nn.functional as F
from funasr.modules.streaming_utils.chunk_utilis import overlap_chunk
from typeguard import check_argument_types
import numpy as np
from funasr.torch_utils.device_funcs import to_device
from funasr.modules.nets_utils import make_pad_mask
from funasr.modules.attention import MultiHeadedAttention, MultiHeadedAttentionSANM, MultiHeadedAttentionSANMwithMask
from funasr.modules.embedding import SinusoidalPositionEncoder, StreamSinusoidalPositionEncoder
@ -349,6 +351,23 @@ class SANMEncoder(AbsEncoder):
return (xs_pad, intermediate_outs), olens, None
return xs_pad, olens, None
def _add_overlap_chunk(self, feats: np.ndarray, cache: dict = {}):
if len(cache) == 0:
return feats
# process last chunk
cache["feats"] = to_device(cache["feats"], device=feats.device)
overlap_feats = torch.cat((cache["feats"], feats), dim=1)
if cache["is_final"]:
cache["feats"] = overlap_feats[:, -cache["chunk_size"][0]:, :]
if not cache["last_chunk"]:
padding_length = sum(cache["chunk_size"]) - overlap_feats.shape[1]
overlap_feats = overlap_feats.transpose(1, 2)
overlap_feats = F.pad(overlap_feats, (0, padding_length))
overlap_feats = overlap_feats.transpose(1, 2)
else:
cache["feats"] = overlap_feats[:, -(cache["chunk_size"][0] + cache["chunk_size"][2]):, :]
return overlap_feats
def forward_chunk(self,
xs_pad: torch.Tensor,
ilens: torch.Tensor,
@ -360,7 +379,7 @@ class SANMEncoder(AbsEncoder):
xs_pad = xs_pad
else:
xs_pad = self.embed(xs_pad, cache)
xs_pad = self._add_overlap_chunk(xs_pad, cache)
encoder_outs = self.encoders0(xs_pad, None, None, None, None)
xs_pad, masks = encoder_outs[0], encoder_outs[1]
intermediate_outs = []

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@ -2,6 +2,7 @@ import torch
from torch import nn
import logging
import numpy as np
from funasr.torch_utils.device_funcs import to_device
from funasr.modules.nets_utils import make_pad_mask
from funasr.modules.streaming_utils.utils import sequence_mask
@ -200,7 +201,7 @@ class CifPredictorV2(nn.Module):
return acoustic_embeds, token_num, alphas, cif_peak
def forward_chunk(self, hidden, cache=None):
b, t, d = hidden.size()
batch_size, len_time, hidden_size = hidden.shape
h = hidden
context = h.transpose(1, 2)
queries = self.pad(context)
@ -211,58 +212,81 @@ class CifPredictorV2(nn.Module):
alphas = torch.nn.functional.relu(alphas * self.smooth_factor - self.noise_threshold)
alphas = alphas.squeeze(-1)
mask_chunk_predictor = None
if cache is not None:
mask_chunk_predictor = None
mask_chunk_predictor = torch.zeros_like(alphas)
mask_chunk_predictor[:, cache["pad_left"]:cache["stride"] + cache["pad_left"]] = 1.0
if mask_chunk_predictor is not None:
alphas = alphas * mask_chunk_predictor
if cache is not None:
if cache["is_final"]:
alphas[:, cache["stride"] + cache["pad_left"] - 1] += 0.45
if cache["cif_hidden"] is not None:
hidden = torch.cat((cache["cif_hidden"], hidden), 1)
if cache["cif_alphas"] is not None:
alphas = torch.cat((cache["cif_alphas"], alphas), -1)
token_num = alphas.sum(-1)
acoustic_embeds, cif_peak = cif(hidden, alphas, self.threshold)
len_time = alphas.size(-1)
last_fire_place = len_time - 1
last_fire_remainds = 0.0
pre_alphas_length = 0
last_fire = False
mask_chunk_peak_predictor = None
if cache is not None:
mask_chunk_peak_predictor = None
mask_chunk_peak_predictor = torch.zeros_like(cif_peak)
if cache["cif_alphas"] is not None:
pre_alphas_length = cache["cif_alphas"].size(-1)
mask_chunk_peak_predictor[:, :pre_alphas_length] = 1.0
mask_chunk_peak_predictor[:, pre_alphas_length + cache["pad_left"]:pre_alphas_length + cache["stride"] + cache["pad_left"]] = 1.0
if mask_chunk_peak_predictor is not None:
cif_peak = cif_peak * mask_chunk_peak_predictor.squeeze(-1)
for i in range(len_time):
if cif_peak[0][len_time - 1 - i] > self.threshold or cif_peak[0][len_time - 1 - i] == self.threshold:
last_fire_place = len_time - 1 - i
last_fire_remainds = cif_peak[0][len_time - 1 - i] - self.threshold
last_fire = True
break
if last_fire:
last_fire_remainds = torch.tensor([last_fire_remainds], dtype=alphas.dtype).to(alphas.device)
cache["cif_hidden"] = hidden[:, last_fire_place:, :]
cache["cif_alphas"] = torch.cat((last_fire_remainds.unsqueeze(0), alphas[:, last_fire_place+1:]), -1)
else:
cache["cif_hidden"] = hidden
cache["cif_alphas"] = alphas
token_num_int = token_num.floor().type(torch.int32).item()
return acoustic_embeds[:, 0:token_num_int, :], token_num, alphas, cif_peak
token_length = []
list_fires = []
list_frames = []
cache_alphas = []
cache_hiddens = []
if cache is not None and "chunk_size" in cache:
alphas[:, :cache["chunk_size"][0]] = 0.0
alphas[:, sum(cache["chunk_size"][:2]):] = 0.0
if cache is not None and "cif_alphas" in cache and "cif_hidden" in cache:
cache["cif_hidden"] = to_device(cache["cif_hidden"], device=hidden.device)
cache["cif_alphas"] = to_device(cache["cif_alphas"], device=alphas.device)
hidden = torch.cat((cache["cif_hidden"], hidden), dim=1)
alphas = torch.cat((cache["cif_alphas"], alphas), dim=1)
if cache is not None and "last_chunk" in cache and cache["last_chunk"]:
tail_hidden = torch.zeros((batch_size, 1, hidden_size), device=hidden.device)
tail_alphas = torch.tensor([[self.tail_threshold]], device=alphas.device)
tail_alphas = torch.tile(tail_alphas, (batch_size, 1))
hidden = torch.cat((hidden, tail_hidden), dim=1)
alphas = torch.cat((alphas, tail_alphas), dim=1)
len_time = alphas.shape[1]
for b in range(batch_size):
integrate = 0.0
frames = torch.zeros((hidden_size), device=hidden.device)
list_frame = []
list_fire = []
for t in range(len_time):
alpha = alphas[b][t]
if alpha + integrate < self.threshold:
integrate += alpha
list_fire.append(integrate)
frames += alpha * hidden[b][t]
else:
frames += (self.threshold - integrate) * hidden[b][t]
list_frame.append(frames)
integrate += alpha
list_fire.append(integrate)
integrate -= self.threshold
frames = integrate * hidden[b][t]
cache_alphas.append(integrate)
if integrate > 0.0:
cache_hiddens.append(frames / integrate)
else:
cache_hiddens.append(frames)
token_length.append(torch.tensor(len(list_frame), device=alphas.device))
list_fires.append(list_fire)
list_frames.append(list_frame)
cache["cif_alphas"] = torch.stack(cache_alphas, axis=0)
cache["cif_alphas"] = torch.unsqueeze(cache["cif_alphas"], axis=0)
cache["cif_hidden"] = torch.stack(cache_hiddens, axis=0)
cache["cif_hidden"] = torch.unsqueeze(cache["cif_hidden"], axis=0)
max_token_len = max(token_length)
if max_token_len == 0:
return hidden, torch.stack(token_length, 0)
list_ls = []
for b in range(batch_size):
pad_frames = torch.zeros((max_token_len - token_length[b], hidden_size), device=alphas.device)
if token_length[b] == 0:
list_ls.append(pad_frames)
else:
list_frames[b] = torch.stack(list_frames[b])
list_ls.append(torch.cat((list_frames[b], pad_frames), dim=0))
cache["cif_alphas"] = torch.stack(cache_alphas, axis=0)
cache["cif_alphas"] = torch.unsqueeze(cache["cif_alphas"], axis=0)
cache["cif_hidden"] = torch.stack(cache_hiddens, axis=0)
cache["cif_hidden"] = torch.unsqueeze(cache["cif_hidden"], axis=0)
return torch.stack(list_ls, 0), torch.stack(token_length, 0)
def tail_process_fn(self, hidden, alphas, token_num=None, mask=None):
b, t, d = hidden.size()

View File

@ -425,21 +425,14 @@ class StreamSinusoidalPositionEncoder(torch.nn.Module):
return encoding.type(dtype)
def forward(self, x, cache=None):
start_idx = 0
pad_left = 0
pad_right = 0
batch_size, timesteps, input_dim = x.size()
start_idx = 0
if cache is not None:
start_idx = cache["start_idx"]
pad_left = cache["left"]
pad_right = cache["right"]
cache["start_idx"] += timesteps
positions = torch.arange(1, timesteps+start_idx+1)[None, :]
position_encoding = self.encode(positions, input_dim, x.dtype).to(x.device)
outputs = x + position_encoding[:, start_idx: start_idx + timesteps]
outputs = outputs.transpose(1, 2)
outputs = F.pad(outputs, (pad_left, pad_right))
outputs = outputs.transpose(1, 2)
return outputs
return x + position_encoding[:, start_idx: start_idx + timesteps]
class StreamingRelPositionalEncoding(torch.nn.Module):
"""Relative positional encoding.