This commit is contained in:
游雁 2023-04-14 10:24:13 +08:00
parent 06ffcc7596
commit f0fdc051fb
22 changed files with 179 additions and 161 deletions

View File

@ -10,7 +10,7 @@ from funasr.export.models.encoder.sanm_encoder import SANMVadEncoder as SANMVadE
class CT_Transformer(nn.Module):
"""
Author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
CT-Transformer: Controllable time-delay transformer for real-time punctuation prediction and disfluency detection
https://arxiv.org/pdf/2003.01309.pdf
"""
@ -81,7 +81,7 @@ class CT_Transformer(nn.Module):
class CT_Transformer_VadRealtime(nn.Module):
"""
Author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
CT-Transformer: Controllable time-delay transformer for real-time punctuation prediction and disfluency detection
https://arxiv.org/pdf/2003.01309.pdf
"""

View File

@ -19,7 +19,7 @@ from funasr.export.models.decoder.transformer_decoder import ParaformerDecoderSA
class Paraformer(nn.Module):
"""
Author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
Paraformer: Fast and Accurate Parallel Transformer for Non-autoregressive End-to-End Speech Recognition
https://arxiv.org/abs/2206.08317
"""
@ -112,7 +112,7 @@ class Paraformer(nn.Module):
class BiCifParaformer(nn.Module):
"""
Author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
Paraformer: Fast and Accurate Parallel Transformer for Non-autoregressive End-to-End Speech Recognition
https://arxiv.org/abs/2206.08317
"""

View File

@ -102,7 +102,7 @@ class ContextualBiasDecoder(nn.Module):
class ContextualParaformerDecoder(ParaformerSANMDecoder):
"""
author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
Paraformer: Fast and Accurate Parallel Transformer for Non-autoregressive End-to-End Speech Recognition
https://arxiv.org/abs/2006.01713
"""

View File

@ -151,7 +151,7 @@ class DecoderLayerSANM(nn.Module):
class FsmnDecoderSCAMAOpt(BaseTransformerDecoder):
"""
author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
SCAMA: Streaming chunk-aware multihead attention for online end-to-end speech recognition
https://arxiv.org/abs/2006.01713
@ -812,7 +812,7 @@ class FsmnDecoderSCAMAOpt(BaseTransformerDecoder):
class ParaformerSANMDecoder(BaseTransformerDecoder):
"""
author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
Paraformer: Fast and Accurate Parallel Transformer for Non-autoregressive End-to-End Speech Recognition
https://arxiv.org/abs/2006.01713
"""

View File

@ -405,7 +405,7 @@ class TransformerDecoder(BaseTransformerDecoder):
class ParaformerDecoderSAN(BaseTransformerDecoder):
"""
author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
Paraformer: Fast and Accurate Parallel Transformer for Non-autoregressive End-to-End Speech Recognition
https://arxiv.org/abs/2006.01713
"""

View File

@ -44,7 +44,7 @@ else:
class Paraformer(AbsESPnetModel):
"""
Author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
Paraformer: Fast and Accurate Parallel Transformer for Non-autoregressive End-to-End Speech Recognition
https://arxiv.org/abs/2206.08317
"""
@ -612,7 +612,7 @@ class Paraformer(AbsESPnetModel):
class ParaformerBert(Paraformer):
"""
Author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
Paraformer2: advanced paraformer with LFMMI and bert for non-autoregressive end-to-end speech recognition
"""

View File

@ -32,7 +32,7 @@ else:
class TimestampPredictor(AbsESPnetModel):
"""
Author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
"""
def __init__(

View File

@ -40,7 +40,7 @@ else:
class UniASR(AbsESPnetModel):
"""
Author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
"""
def __init__(

View File

@ -35,6 +35,11 @@ class VadDetectMode(Enum):
class VADXOptions:
"""
Author: Speech Lab of DAMO Academy, Alibaba Group
Deep-FSMN for Large Vocabulary Continuous Speech Recognition
https://arxiv.org/abs/1803.05030
"""
def __init__(
self,
sample_rate: int = 16000,
@ -99,6 +104,11 @@ class VADXOptions:
class E2EVadSpeechBufWithDoa(object):
"""
Author: Speech Lab of DAMO Academy, Alibaba Group
Deep-FSMN for Large Vocabulary Continuous Speech Recognition
https://arxiv.org/abs/1803.05030
"""
def __init__(self):
self.start_ms = 0
self.end_ms = 0
@ -117,6 +127,11 @@ class E2EVadSpeechBufWithDoa(object):
class E2EVadFrameProb(object):
"""
Author: Speech Lab of DAMO Academy, Alibaba Group
Deep-FSMN for Large Vocabulary Continuous Speech Recognition
https://arxiv.org/abs/1803.05030
"""
def __init__(self):
self.noise_prob = 0.0
self.speech_prob = 0.0
@ -126,6 +141,11 @@ class E2EVadFrameProb(object):
class WindowDetector(object):
"""
Author: Speech Lab of DAMO Academy, Alibaba Group
Deep-FSMN for Large Vocabulary Continuous Speech Recognition
https://arxiv.org/abs/1803.05030
"""
def __init__(self, window_size_ms: int, sil_to_speech_time: int,
speech_to_sil_time: int, frame_size_ms: int):
self.window_size_ms = window_size_ms
@ -192,6 +212,11 @@ class WindowDetector(object):
class E2EVadModel(nn.Module):
"""
Author: Speech Lab of DAMO Academy, Alibaba Group
Deep-FSMN for Large Vocabulary Continuous Speech Recognition
https://arxiv.org/abs/1803.05030
"""
def __init__(self, encoder: FSMN, vad_post_args: Dict[str, Any], frontend=None):
super(E2EVadModel, self).__init__()
self.vad_opts = VADXOptions(**vad_post_args)

View File

@ -67,7 +67,7 @@ class EncoderLayer(nn.Module):
class ConvEncoder(AbsEncoder):
"""
author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
Convolution encoder in OpenNMT framework
"""

View File

@ -117,7 +117,7 @@ class EncoderLayer(nn.Module):
class SelfAttentionEncoder(AbsEncoder):
"""
author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
Self attention encoder in OpenNMT framework
"""

View File

@ -117,7 +117,7 @@ class EncoderLayerSANM(nn.Module):
class SANMEncoder(AbsEncoder):
"""
author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
San-m: Memory equipped self-attention for end-to-end speech recognition
https://arxiv.org/abs/2006.01713
@ -549,7 +549,7 @@ class SANMEncoder(AbsEncoder):
class SANMEncoderChunkOpt(AbsEncoder):
"""
author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
SCAMA: Streaming chunk-aware multihead attention for online end-to-end speech recognition
https://arxiv.org/abs/2006.01713
@ -962,7 +962,7 @@ class SANMEncoderChunkOpt(AbsEncoder):
class SANMVadEncoder(AbsEncoder):
"""
author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
"""

View File

@ -14,7 +14,7 @@ from funasr.train.abs_model import AbsPunctuation
class TargetDelayTransformer(AbsPunctuation):
"""
Author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
CT-Transformer: Controllable time-delay transformer for real-time punctuation prediction and disfluency detection
https://arxiv.org/pdf/2003.01309.pdf
"""

View File

@ -12,7 +12,7 @@ from funasr.train.abs_model import AbsPunctuation
class VadRealtimeTransformer(AbsPunctuation):
"""
Author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
CT-Transformer: Controllable time-delay transformer for real-time punctuation prediction and disfluency detection
https://arxiv.org/pdf/2003.01309.pdf
"""

View File

@ -11,7 +11,7 @@ from funasr.modules.streaming_utils.utils import sequence_mask
class overlap_chunk():
"""
author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
San-m: Memory equipped self-attention for end-to-end speech recognition
https://arxiv.org/abs/2006.01713

View File

@ -1,5 +1,5 @@
import soundfile
from funasr_onnx.vad_bin import Fsmn_vad
from funasr_onnx import Fsmn_vad
model_dir = "/mnt/ailsa.zly/tfbase/espnet_work/FunASR_dev_zly/export/damo/speech_fsmn_vad_zh-cn-16k-common-pytorch"

View File

@ -1,10 +1,10 @@
import soundfile
from funasr_onnx.vad_online_bin import Fsmn_vad
from funasr_onnx import Fsmn_vad_online
model_dir = "/mnt/ailsa.zly/tfbase/espnet_work/FunASR_dev_zly/export/damo/speech_fsmn_vad_zh-cn-16k-common-pytorch"
wav_path = "/mnt/ailsa.zly/tfbase/espnet_work/FunASR_dev_zly/egs_modelscope/vad/speech_fsmn_vad_zh-cn-16k-common/vad_example_16k.wav"
model = Fsmn_vad(model_dir)
model = Fsmn_vad_online(model_dir)
##online vad

View File

@ -1,5 +1,6 @@
# -*- encoding: utf-8 -*-
from .paraformer_bin import Paraformer
from .vad_bin import Fsmn_vad
from .vad_bin import Fsmn_vad_online
from .punc_bin import CT_Transformer
from .punc_bin import CT_Transformer_VadRealtime

View File

@ -14,7 +14,7 @@ logging = get_logger()
class CT_Transformer():
"""
Author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
CT-Transformer: Controllable time-delay transformer for real-time punctuation prediction and disfluency detection
https://arxiv.org/pdf/2003.01309.pdf
"""
@ -125,7 +125,7 @@ class CT_Transformer():
class CT_Transformer_VadRealtime(CT_Transformer):
"""
Author: Speech Lab, Alibaba Group, China
Author: Speech Lab of DAMO Academy, Alibaba Group
CT-Transformer: Controllable time-delay transformer for real-time punctuation prediction and disfluency detection
https://arxiv.org/pdf/2003.01309.pdf
"""

View File

@ -11,13 +11,18 @@ import numpy as np
from .utils.utils import (ONNXRuntimeError,
OrtInferSession, get_logger,
read_yaml)
from .utils.frontend import WavFrontend
from .utils.frontend import WavFrontend, WavFrontendOnline
from .utils.e2e_vad import E2EVadModel
logging = get_logger()
class Fsmn_vad():
"""
Author: Speech Lab of DAMO Academy, Alibaba Group
Deep-FSMN for Large Vocabulary Continuous Speech Recognition
https://arxiv.org/abs/1803.05030
"""
def __init__(self, model_dir: Union[str, Path] = None,
batch_size: int = 1,
device_id: Union[str, int] = "-1",
@ -152,3 +157,124 @@ class Fsmn_vad():
scores, out_caches = outputs[0], outputs[1:]
return scores, out_caches
class Fsmn_vad_online():
"""
Author: Speech Lab of DAMO Academy, Alibaba Group
Deep-FSMN for Large Vocabulary Continuous Speech Recognition
https://arxiv.org/abs/1803.05030
"""
def __init__(self, model_dir: Union[str, Path] = None,
batch_size: int = 1,
device_id: Union[str, int] = "-1",
quantize: bool = False,
intra_op_num_threads: int = 4,
max_end_sil: int = None,
):
if not Path(model_dir).exists():
raise FileNotFoundError(f'{model_dir} does not exist.')
model_file = os.path.join(model_dir, 'model.onnx')
if quantize:
model_file = os.path.join(model_dir, 'model_quant.onnx')
config_file = os.path.join(model_dir, 'vad.yaml')
cmvn_file = os.path.join(model_dir, 'vad.mvn')
config = read_yaml(config_file)
self.frontend = WavFrontendOnline(
cmvn_file=cmvn_file,
**config['frontend_conf']
)
self.ort_infer = OrtInferSession(model_file, device_id, intra_op_num_threads=intra_op_num_threads)
self.batch_size = batch_size
self.vad_scorer = E2EVadModel(config["vad_post_conf"])
self.max_end_sil = max_end_sil if max_end_sil is not None else config["vad_post_conf"]["max_end_silence_time"]
self.encoder_conf = config["encoder_conf"]
def prepare_cache(self, in_cache: list = []):
if len(in_cache) > 0:
return in_cache
fsmn_layers = self.encoder_conf["fsmn_layers"]
proj_dim = self.encoder_conf["proj_dim"]
lorder = self.encoder_conf["lorder"]
for i in range(fsmn_layers):
cache = np.zeros((1, proj_dim, lorder - 1, 1)).astype(np.float32)
in_cache.append(cache)
return in_cache
def __call__(self, audio_in: np.ndarray, **kwargs) -> List:
waveforms = np.expand_dims(audio_in, axis=0)
param_dict = kwargs.get('param_dict', dict())
is_final = param_dict.get('is_final', False)
feats, feats_len = self.extract_feat(waveforms, is_final)
segments = []
if feats.size != 0:
in_cache = param_dict.get('in_cache', list())
in_cache = self.prepare_cache(in_cache)
try:
inputs = [feats]
inputs.extend(in_cache)
scores, out_caches = self.infer(inputs)
param_dict['in_cache'] = out_caches
waveforms = self.frontend.get_waveforms()
segments = self.vad_scorer(scores, waveforms, is_final=is_final, max_end_sil=self.max_end_sil,
online=True)
except ONNXRuntimeError:
# logging.warning(traceback.format_exc())
logging.warning("input wav is silence or noise")
segments = []
return segments
def load_data(self,
wav_content: Union[str, np.ndarray, List[str]], fs: int = None) -> List:
def load_wav(path: str) -> np.ndarray:
waveform, _ = librosa.load(path, sr=fs)
return waveform
if isinstance(wav_content, np.ndarray):
return [wav_content]
if isinstance(wav_content, str):
return [load_wav(wav_content)]
if isinstance(wav_content, list):
return [load_wav(path) for path in wav_content]
raise TypeError(
f'The type of {wav_content} is not in [str, np.ndarray, list]')
def extract_feat(self,
waveforms: np.ndarray, is_final: bool = False
) -> Tuple[np.ndarray, np.ndarray]:
waveforms_lens = np.zeros(waveforms.shape[0]).astype(np.int32)
for idx, waveform in enumerate(waveforms):
waveforms_lens[idx] = waveform.shape[-1]
feats, feats_len = self.frontend.extract_fbank(waveforms, waveforms_lens, is_final)
# feats.append(feat)
# feats_len.append(feat_len)
# feats = self.pad_feats(feats, np.max(feats_len))
# feats_len = np.array(feats_len).astype(np.int32)
return feats.astype(np.float32), feats_len.astype(np.int32)
@staticmethod
def pad_feats(feats: List[np.ndarray], max_feat_len: int) -> np.ndarray:
def pad_feat(feat: np.ndarray, cur_len: int) -> np.ndarray:
pad_width = ((0, max_feat_len - cur_len), (0, 0))
return np.pad(feat, pad_width, 'constant', constant_values=0)
feat_res = [pad_feat(feat, feat.shape[0]) for feat in feats]
feats = np.array(feat_res).astype(np.float32)
return feats
def infer(self, feats: List) -> Tuple[np.ndarray, np.ndarray]:
outputs = self.ort_infer(feats)
scores, out_caches = outputs[0], outputs[1:]
return scores, out_caches

View File

@ -1,134 +0,0 @@
# -*- encoding: utf-8 -*-
import os.path
from pathlib import Path
from typing import List, Union, Tuple
import copy
import librosa
import numpy as np
from .utils.utils import (ONNXRuntimeError,
OrtInferSession, get_logger,
read_yaml)
from .utils.frontend import WavFrontendOnline
from .utils.e2e_vad import E2EVadModel
logging = get_logger()
class Fsmn_vad():
def __init__(self, model_dir: Union[str, Path] = None,
batch_size: int = 1,
device_id: Union[str, int] = "-1",
quantize: bool = False,
intra_op_num_threads: int = 4,
max_end_sil: int = None,
):
if not Path(model_dir).exists():
raise FileNotFoundError(f'{model_dir} does not exist.')
model_file = os.path.join(model_dir, 'model.onnx')
if quantize:
model_file = os.path.join(model_dir, 'model_quant.onnx')
config_file = os.path.join(model_dir, 'vad.yaml')
cmvn_file = os.path.join(model_dir, 'vad.mvn')
config = read_yaml(config_file)
self.frontend = WavFrontendOnline(
cmvn_file=cmvn_file,
**config['frontend_conf']
)
self.ort_infer = OrtInferSession(model_file, device_id, intra_op_num_threads=intra_op_num_threads)
self.batch_size = batch_size
self.vad_scorer = E2EVadModel(config["vad_post_conf"])
self.max_end_sil = max_end_sil if max_end_sil is not None else config["vad_post_conf"]["max_end_silence_time"]
self.encoder_conf = config["encoder_conf"]
def prepare_cache(self, in_cache: list = []):
if len(in_cache) > 0:
return in_cache
fsmn_layers = self.encoder_conf["fsmn_layers"]
proj_dim = self.encoder_conf["proj_dim"]
lorder = self.encoder_conf["lorder"]
for i in range(fsmn_layers):
cache = np.zeros((1, proj_dim, lorder-1, 1)).astype(np.float32)
in_cache.append(cache)
return in_cache
def __call__(self, audio_in: np.ndarray, **kwargs) -> List:
waveforms = np.expand_dims(audio_in, axis=0)
param_dict = kwargs.get('param_dict', dict())
is_final = param_dict.get('is_final', False)
feats, feats_len = self.extract_feat(waveforms, is_final)
segments = []
if feats.size != 0:
in_cache = param_dict.get('in_cache', list())
in_cache = self.prepare_cache(in_cache)
try:
inputs = [feats]
inputs.extend(in_cache)
scores, out_caches = self.infer(inputs)
param_dict['in_cache'] = out_caches
waveforms = self.frontend.get_waveforms()
segments = self.vad_scorer(scores, waveforms, is_final=is_final, max_end_sil=self.max_end_sil, online=True)
except ONNXRuntimeError:
logging.warning(traceback.format_exc())
logging.warning("input wav is silence or noise")
segments = []
return segments
def load_data(self,
wav_content: Union[str, np.ndarray, List[str]], fs: int = None) -> List:
def load_wav(path: str) -> np.ndarray:
waveform, _ = librosa.load(path, sr=fs)
return waveform
if isinstance(wav_content, np.ndarray):
return [wav_content]
if isinstance(wav_content, str):
return [load_wav(wav_content)]
if isinstance(wav_content, list):
return [load_wav(path) for path in wav_content]
raise TypeError(
f'The type of {wav_content} is not in [str, np.ndarray, list]')
def extract_feat(self,
waveforms: np.ndarray, is_final: bool = False
) -> Tuple[np.ndarray, np.ndarray]:
waveforms_lens = np.zeros(waveforms.shape[0]).astype(np.int32)
for idx, waveform in enumerate(waveforms):
waveforms_lens[idx] = waveform.shape[-1]
feats, feats_len = self.frontend.extract_fbank(waveforms, waveforms_lens, is_final)
# feats.append(feat)
# feats_len.append(feat_len)
# feats = self.pad_feats(feats, np.max(feats_len))
# feats_len = np.array(feats_len).astype(np.int32)
return feats.astype(np.float32), feats_len.astype(np.int32)
@staticmethod
def pad_feats(feats: List[np.ndarray], max_feat_len: int) -> np.ndarray:
def pad_feat(feat: np.ndarray, cur_len: int) -> np.ndarray:
pad_width = ((0, max_feat_len - cur_len), (0, 0))
return np.pad(feat, pad_width, 'constant', constant_values=0)
feat_res = [pad_feat(feat, feat.shape[0]) for feat in feats]
feats = np.array(feat_res).astype(np.float32)
return feats
def infer(self, feats: List) -> Tuple[np.ndarray, np.ndarray]:
outputs = self.ort_infer(feats)
scores, out_caches = outputs[0], outputs[1:]
return scores, out_caches

View File

@ -13,7 +13,7 @@ def get_readme():
MODULE_NAME = 'funasr_onnx'
VERSION_NUM = '0.0.3'
VERSION_NUM = '0.0.4'
setuptools.setup(
name=MODULE_NAME,