ar_asr
Author: Heli Qi Affiliation: NAIST Date: 2022.07
ARASR
Bases: Model
Auto-Regressive Attention-based Automatic Speech Recognition (AR-ASR) implementation.
The neural network structure of an ASR
Model object is made up of 3 Module members:
-
an
ASREncoder
made up of:frontend
converts the raw waveforms into acoustic features on-the-fly.normalize
normalizes the extracted acoustic features to normal distribution for faster convergence.specaug
randomly warps and masks the normalized acoustic features.prenet
preprocesses the augmented acoustic features before passing them to the encoder.encoder
extracts the encoder hidden representations of the preprocessed acoustic features and passes them toARASRDecoder
.
-
an
ARASRDecoder
made up of:embedding
embeds each tokens in the input sentence into token embedding vectors.decoder
extracts the decoder hidden representations based on the token embedding vectors and encoder hidden representations.postnet
predicts the probability of the next tokens by the decoder hidden representations.
-
(optional) a CTC layer made up of a 'TokenPostnet'
Source code in speechain/model/ar_asr.py
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criterion_init(ce_loss=None, ilm_loss=None, ctc_loss=None, att_guid_loss=None)
This function initializes all the necessary Criterion members
speechain.criterion.cross_entropy.CrossEntropy
for training loss calculation.speechain.criterion.ctc.CTCLoss
for training loss calculation.speechain.criterion.accuracy.Accuracy
for teacher-forcing validation accuracy calculation.speechain.criterion.error_rate.ErrorRate
for evaluation CER & WER calculation.
Parameters:
Name | Type | Description | Default |
---|---|---|---|
ce_loss
|
Dict[str, Any]
|
Dict[str, Any] The arguments for CrossEntropy(). If not given, the default setting of CrossEntropy() will be used. Please refer to speechain.criterion.cross_entropy.CrossEntropy for more details. |
None
|
ilm_loss
|
Dict[str, Any]
|
|
None
|
ctc_loss
|
Dict[str, Any] or bool
|
Dict[str, Any] or bool The arguments for CTCLoss(). If not given, self.ctc_loss won't be initialized. This argument can also be set to a bool value 'True'. If True, the default setting of CTCLoss() will be used. Please refer to speechain.criterion.ctc.CTCLoss for more details. |
None
|
att_guid_loss
|
Dict[str, Any] or bool
|
Dict[str, Any] or bool The arguments for AttentionGuidance(). If not given, self.att_guid_loss won't be initialized. This argument can also be set to a bool value 'True'. If True, the default setting of AttentionGuidance() will be used. Please refer to speechain.criterion.att_guid.AttentionGuidance for more details. |
None
|
Source code in speechain/model/ar_asr.py
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inference(infer_conf, feat=None, feat_len=None, text=None, text_len=None, domain=None, return_att=False, decode_only=False, teacher_forcing=False)
The inference function for ASR models. There are two steps in this function: 1. Decode the input speech into hypothesis transcript 2. Evaluate the hypothesis transcript by the ground-truth
This function can be called for model evaluation, on-the-fly model visualization, and even pseudo transcript generation during training.
Parameters:
Name | Type | Description | Default |
---|---|---|---|
feat
|
Tensor
|
torch.Tensor The speech data to be inferred. |
None
|
feat_len
|
Tensor
|
torch.Tensor
The length of |
None
|
text
|
Tensor
|
torch.Tensor The ground-truth transcript for the input speech |
None
|
text_len
|
Tensor
|
torch.Tensor
The length of |
None
|
domain
|
str
|
str = None
This argument indicates which domain the input speech belongs to.
It's used to indicate the |
None
|
return_att
|
bool
|
bool = False Whether the attention matrix of the input speech is returned. |
False
|
decode_only
|
bool
|
bool = False Whether skip the evaluation step and do the decoding step only. |
False
|
teacher_forcing
|
bool
|
bool = True Whether you use the teacher-forcing technique to generate the hypothesis transcript. |
False
|
infer_conf
|
Dict
|
Dict
The inference configuration given from the |
required |
Dict
Type | Description |
---|---|
Dict[str, Any]
|
A Dict containing all the decoding and evaluation results. |
Source code in speechain/model/ar_asr.py
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module_forward(epoch=None, feat=None, text=None, feat_len=None, text_len=None, domain=None, return_att=False, **kwargs)
Parameters:
Name | Type | Description | Default |
---|---|---|---|
feat
|
Tensor
|
(batch, feat_maxlen, feat_dim) The input speech data. feat_dim = 1 in the case of raw speech waveforms. |
None
|
feat_len
|
Tensor
|
(batch,) The lengths of input speech data |
None
|
text
|
Tensor
|
(batch, text_maxlen)
The input text data with |
None
|
text_len
|
Tensor
|
(batch,) The lengths of input text data |
None
|
epoch
|
int
|
int The number of the current training epoch. Mainly used for mean&std calculation in the feature normalization |
None
|
domain
|
str
|
str = None |
None
|
return_att
|
bool
|
bool Controls whether the attention matrices of each layer in the encoder and decoder will be returned. |
False
|
kwargs
|
Temporary register used to store the redundant arguments. |
{}
|
Source code in speechain/model/ar_asr.py
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module_init(token_type, token_path, enc_prenet, encoder, dec_emb, decoder, frontend=None, normalize=None, specaug=None, ilm_weight=0.0, ilm_sub_weight=0.0, ctc_weight=0.0, sample_rate=16000, audio_format='wav', return_att_type=None, return_att_head_num=2, return_att_layer_num=2, lm_model_cfg=None, lm_model_path=None)
This initialization function contains 4 steps:
1. Tokenizer
initialization.
2. ASREncoder
initialization.
3. ARASRDecoder
initialization.
4. (optional) 'CTC' layer initialization
The input arguments of this function are two-fold:
1. the ones from customize_conf
of model
in train_cfg
2. the ones from module_conf
of model
in train_cfg
Parameters:
Name | Type | Description | Default |
---|---|---|---|
frontend
|
Dict
|
(optional)
The configuration of the acoustic feature extraction frontend in the |
None
|
normalize
|
Dict or bool
|
(optional)
The configuration of the normalization layer in the |
None
|
specaug
|
Dict or bool
|
(optional)
The configuration of the SpecAugment layer in the |
None
|
enc_prenet
|
Dict
|
(mandatory)
The configuration of the prenet in the |
required |
encoder
|
Dict
|
(mandatory)
The configuration of the encoder main body in the |
required |
dec_emb
|
Dict
|
(mandatory)
The configuration of the embedding layer in the |
required |
decoder
|
Dict
|
(mandatory)
The configuration of the decoder main body in the |
required |
token_type
|
str
|
(mandatory) The type of the built-in tokenizer. |
required |
token_path
|
str
|
(mandatory) The path of the vocabulary for initializing the built-in tokenizer. |
required |
sample_rate
|
int
|
int = 16000 (optional) The sampling rate of the input speech. Currently, it's used for acoustic feature extraction frontend initialization and tensorboard register of the input speech for model visualization. In the future, this argument will also be used to on-the-fly downsample the input speech. |
16000
|
audio_format
|
str
|
(optional) This argument is only used for input speech recording during model visualization. |
'wav'
|
return_att_type
|
List[str] or str
|
List[str] or str = ['encdec', 'enc', 'dec'] The type of attentions you want to return for both attention guidance and attention visualization. It can be given as a string (one type) or a list of strings (multiple types). The type should be one of 1. 'encdec': the encoder-decoder attention, shared by both Transformer and RNN 2. 'enc': the encoder self-attention, only for Transformer 3. 'dec': the decoder self-attention, only for Transformer |
None
|
return_att_head_num
|
int
|
int = -1 The number of returned attention heads. If -1, all the heads in an attention layer will be returned. RNN can be considered to one-head attention, so return_att_head_num > 1 is equivalent to 1 for RNN. |
2
|
return_att_layer_num
|
int
|
int = 1 The number of returned attention layers. If -1, all the attention layers will be returned. RNN can be considered to one-layer attention, so return_att_layer_num > 1 is equivalent to 1 for RNN. |
2
|
lm_model_cfg
|
Dict or str
|
Dict or str The configuration for the language model used for joint decoding. Can be either a Dict or a string indicating where the .yaml model configuration file is placed. |
None
|
lm_model_path
|
str
|
str The string indicating where the .pth model parameter file is placed. |
None
|
Source code in speechain/model/ar_asr.py
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MultiDataLoaderARASR
Bases: ARASR
Auto-Regressive ASR model trained by multiple dataloaders.
Source code in speechain/model/ar_asr.py
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