Source code for torchaudio.pipelines.rnnt_pipeline
import json
import math
from abc import ABC, abstractmethod
from dataclasses import dataclass
from functools import partial
from typing import Callable, List, Tuple
import torch
import torchaudio
from torchaudio._internal import module_utils
from torchaudio.models import emformer_rnnt_base, RNNT, RNNTBeamSearch
__all__ = []
_decibel = 2 * 20 * math.log10(torch.iinfo(torch.int16).max)
_gain = pow(10, 0.05 * _decibel)
def _piecewise_linear_log(x):
x[x > math.e] = torch.log(x[x > math.e])
x[x <= math.e] = x[x <= math.e] / math.e
return x
class _FunctionalModule(torch.nn.Module):
def __init__(self, functional):
super().__init__()
self.functional = functional
def forward(self, input):
return self.functional(input)
class _GlobalStatsNormalization(torch.nn.Module):
def __init__(self, global_stats_path):
super().__init__()
with open(global_stats_path) as f:
blob = json.loads(f.read())
self.register_buffer("mean", torch.tensor(blob["mean"]))
self.register_buffer("invstddev", torch.tensor(blob["invstddev"]))
def forward(self, input):
return (input - self.mean) * self.invstddev
class _FeatureExtractor(ABC):
@abstractmethod
def __call__(self, input: torch.Tensor) -> Tuple[torch.Tensor, torch.Tensor]:
"""Generates features and length output from the given input tensor.
Args:
input (torch.Tensor): input tensor.
Returns:
(torch.Tensor, torch.Tensor):
torch.Tensor:
Features, with shape `(length, *)`.
torch.Tensor:
Length, with shape `(1,)`.
"""
class _TokenProcessor(ABC):
@abstractmethod
def __call__(self, tokens: List[int], **kwargs) -> str:
"""Decodes given list of tokens to text sequence.
Args:
tokens (List[int]): list of tokens to decode.
Returns:
str:
Decoded text sequence.
"""
class _ModuleFeatureExtractor(torch.nn.Module, _FeatureExtractor):
"""``torch.nn.Module``-based feature extraction pipeline.
Args:
pipeline (torch.nn.Module): module that implements feature extraction logic.
"""
def __init__(self, pipeline: torch.nn.Module) -> None:
super().__init__()
self.pipeline = pipeline
def forward(self, input: torch.Tensor) -> Tuple[torch.Tensor, torch.Tensor]:
"""Generates features and length output from the given input tensor.
Args:
input (torch.Tensor): input tensor.
Returns:
(torch.Tensor, torch.Tensor):
torch.Tensor:
Features, with shape `(length, *)`.
torch.Tensor:
Length, with shape `(1,)`.
"""
features = self.pipeline(input)
length = torch.tensor([features.shape[0]])
return features, length
class _SentencePieceTokenProcessor(_TokenProcessor):
"""SentencePiece-model-based token processor.
Args:
sp_model_path (str): path to SentencePiece model.
"""
def __init__(self, sp_model_path: str) -> None:
if not module_utils.is_module_available("sentencepiece"):
raise RuntimeError("SentencePiece is not available. Please install it.")
import sentencepiece as spm
self.sp_model = spm.SentencePieceProcessor(model_file=sp_model_path)
self.post_process_remove_list = {
self.sp_model.unk_id(),
self.sp_model.eos_id(),
self.sp_model.pad_id(),
}
def __call__(self, tokens: List[int], lstrip: bool = True) -> str:
"""Decodes given list of tokens to text sequence.
Args:
tokens (List[int]): list of tokens to decode.
lstrip (bool, optional): if ``True``, returns text sequence with leading whitespace
removed. (Default: ``True``).
Returns:
str:
Decoded text sequence.
"""
filtered_hypo_tokens = [
token_index for token_index in tokens[1:] if token_index not in self.post_process_remove_list
]
output_string = "".join(self.sp_model.id_to_piece(filtered_hypo_tokens)).replace("\u2581", " ")
if lstrip:
return output_string.lstrip()
else:
return output_string
[docs]@dataclass
class RNNTBundle:
"""torchaudio.pipelines.RNNTBundle()
Dataclass that bundles components for performing automatic speech recognition (ASR, speech-to-text)
inference with an RNN-T model.
More specifically, the class provides methods that produce the featurization pipeline,
decoder wrapping the specified RNN-T model, and output token post-processor that together
constitute a complete end-to-end ASR inference pipeline that produces a text sequence
given a raw waveform.
It can support non-streaming (full-context) inference as well as streaming inference.
Users should not directly instantiate objects of this class; rather, users should use the
instances (representing pre-trained models) that exist within the module,
e.g. :py:obj:`EMFORMER_RNNT_BASE_LIBRISPEECH`.
Example
>>> import torchaudio
>>> from torchaudio.pipelines import EMFORMER_RNNT_BASE_LIBRISPEECH
>>> import torch
>>>
>>> # Non-streaming inference.
>>> # Build feature extractor, decoder with RNN-T model, and token processor.
>>> feature_extractor = EMFORMER_RNNT_BASE_LIBRISPEECH.get_feature_extractor()
100%|███████████████████████████████| 3.81k/3.81k [00:00<00:00, 4.22MB/s]
>>> decoder = EMFORMER_RNNT_BASE_LIBRISPEECH.get_decoder()
Downloading: "https://download.pytorch.org/torchaudio/models/emformer_rnnt_base_librispeech.pt"
100%|███████████████████████████████| 293M/293M [00:07<00:00, 42.1MB/s]
>>> token_processor = EMFORMER_RNNT_BASE_LIBRISPEECH.get_token_processor()
100%|███████████████████████████████| 295k/295k [00:00<00:00, 25.4MB/s]
>>>
>>> # Instantiate LibriSpeech dataset; retrieve waveform for first sample.
>>> dataset = torchaudio.datasets.LIBRISPEECH("/home/librispeech", url="test-clean")
>>> waveform = next(iter(dataset))[0].squeeze()
>>>
>>> with torch.no_grad():
>>> # Produce mel-scale spectrogram features.
>>> features, length = feature_extractor(waveform)
>>>
>>> # Generate top-10 hypotheses.
>>> hypotheses = decoder(features, length, 10)
>>>
>>> # For top hypothesis, convert predicted tokens to text.
>>> text = token_processor(hypotheses[0][0])
>>> print(text)
he hoped there would be stew for dinner turnips and carrots and bruised potatoes and fat mutton pieces to [...]
>>>
>>>
>>> # Streaming inference.
>>> hop_length = EMFORMER_RNNT_BASE_LIBRISPEECH.hop_length
>>> num_samples_segment = EMFORMER_RNNT_BASE_LIBRISPEECH.segment_length * hop_length
>>> num_samples_segment_right_context = (
>>> num_samples_segment + EMFORMER_RNNT_BASE_LIBRISPEECH.right_context_length * hop_length
>>> )
>>>
>>> # Build streaming inference feature extractor.
>>> streaming_feature_extractor = EMFORMER_RNNT_BASE_LIBRISPEECH.get_streaming_feature_extractor()
>>>
>>> # Process same waveform as before, this time sequentially across overlapping segments
>>> # to simulate streaming inference. Note the usage of ``streaming_feature_extractor`` and ``decoder.infer``.
>>> state, hypothesis = None, None
>>> for idx in range(0, len(waveform), num_samples_segment):
>>> segment = waveform[idx: idx + num_samples_segment_right_context]
>>> segment = torch.nn.functional.pad(segment, (0, num_samples_segment_right_context - len(segment)))
>>> with torch.no_grad():
>>> features, length = streaming_feature_extractor(segment)
>>> hypotheses, state = decoder.infer(features, length, 10, state=state, hypothesis=hypothesis)
>>> hypothesis = hypotheses[0]
>>> transcript = token_processor(hypothesis[0])
>>> if transcript:
>>> print(transcript, end=" ", flush=True)
he hoped there would be stew for dinner turn ips and car rots and bru 'd oes and fat mut ton pieces to [...]
"""
_rnnt_path: str
_rnnt_factory_func: Callable[[], RNNT]
_global_stats_path: str
_sp_model_path: str
_right_padding: int
_blank: int
_sample_rate: int
_n_fft: int
_n_mels: int
_hop_length: int
_segment_length: int
_right_context_length: int
def _get_model(self) -> RNNT:
model = self._rnnt_factory_func()
path = torchaudio.utils.download_asset(self._rnnt_path)
state_dict = torch.load(path)
model.load_state_dict(state_dict)
model.eval()
return model
@property
def sample_rate(self) -> int:
"""Sample rate (in cycles per second) of input waveforms.
:type: int
"""
return self._sample_rate
@property
def n_fft(self) -> int:
"""Size of FFT window to use.
:type: int
"""
return self._n_fft
@property
def n_mels(self) -> int:
"""Number of mel spectrogram features to extract from input waveforms.
:type: int
"""
return self._n_mels
@property
def hop_length(self) -> int:
"""Number of samples between successive frames in input expected by model.
:type: int
"""
return self._hop_length
@property
def segment_length(self) -> int:
"""Number of frames in segment in input expected by model.
:type: int
"""
return self._segment_length
@property
def right_context_length(self) -> int:
"""Number of frames in right contextual block in input expected by model.
:type: int
"""
return self._right_context_length
[docs] def get_decoder(self) -> RNNTBeamSearch:
"""Constructs RNN-T decoder.
Returns:
RNNTBeamSearch
"""
model = self._get_model()
return RNNTBeamSearch(model, self._blank)
[docs] def get_feature_extractor(self) -> FeatureExtractor:
"""Constructs feature extractor for non-streaming (full-context) ASR.
Returns:
FeatureExtractor
"""
local_path = torchaudio.utils.download_asset(self._global_stats_path)
return _ModuleFeatureExtractor(
torch.nn.Sequential(
torchaudio.transforms.MelSpectrogram(
sample_rate=self.sample_rate, n_fft=self.n_fft, n_mels=self.n_mels, hop_length=self.hop_length
),
_FunctionalModule(lambda x: x.transpose(1, 0)),
_FunctionalModule(lambda x: _piecewise_linear_log(x * _gain)),
_GlobalStatsNormalization(local_path),
_FunctionalModule(lambda x: torch.nn.functional.pad(x, (0, 0, 0, self._right_padding))),
)
)
[docs] def get_streaming_feature_extractor(self) -> FeatureExtractor:
"""Constructs feature extractor for streaming (simultaneous) ASR.
Returns:
FeatureExtractor
"""
local_path = torchaudio.utils.download_asset(self._global_stats_path)
return _ModuleFeatureExtractor(
torch.nn.Sequential(
torchaudio.transforms.MelSpectrogram(
sample_rate=self.sample_rate, n_fft=self.n_fft, n_mels=self.n_mels, hop_length=self.hop_length
),
_FunctionalModule(lambda x: x.transpose(1, 0)),
_FunctionalModule(lambda x: _piecewise_linear_log(x * _gain)),
_GlobalStatsNormalization(local_path),
)
)
[docs] def get_token_processor(self) -> TokenProcessor:
"""Constructs token processor.
Returns:
TokenProcessor
"""
local_path = torchaudio.utils.download_asset(self._sp_model_path)
return _SentencePieceTokenProcessor(local_path)
EMFORMER_RNNT_BASE_LIBRISPEECH = RNNTBundle(
_rnnt_path="models/emformer_rnnt_base_librispeech.pt",
_rnnt_factory_func=partial(emformer_rnnt_base, num_symbols=4097),
_global_stats_path="pipeline-assets/global_stats_rnnt_librispeech.json",
_sp_model_path="pipeline-assets/spm_bpe_4096_librispeech.model",
_right_padding=4,
_blank=4096,
_sample_rate=16000,
_n_fft=400,
_n_mels=80,
_hop_length=160,
_segment_length=16,
_right_context_length=4,
)
EMFORMER_RNNT_BASE_LIBRISPEECH.__doc__ = """Pre-trained Emformer-RNNT-based ASR pipeline capable of performing both streaming and non-streaming inference.
The underlying model is constructed by :py:func:`torchaudio.models.emformer_rnnt_base`
and utilizes weights trained on LibriSpeech using training script ``train.py``
`here <https://github.com/pytorch/audio/tree/main/examples/asr/emformer_rnnt>`__ with default arguments.
Please refer to :py:class:`RNNTBundle` for usage instructions.
"""
