Text-to-Speech
ONNX
GGUF
speech-translation
streaming-speech-translation
speech
audio
speech-recognition
automatic-speech-recognition
streaming-asr
ASR
NeMo
ONNX
cache-aware ASR
FastConformer
RNNT
Parakeet
neural-machine-translation
NMT
gemma3
llama-cpp
GGUF
conversational
TTS
xtts
xttsv2
voice-clone
gpt2
hifigan
multilingual
vq
perceiver-encoder
websocket
File size: 17,765 Bytes
f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 0c397a9 f724af4 0c397a9 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 0c397a9 f724af4 0c397a9 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 da63a34 f724af4 | 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 275 276 277 278 279 280 281 282 283 284 285 286 287 288 289 290 291 292 293 294 295 296 297 298 299 300 301 302 303 304 305 306 307 308 309 310 311 312 313 314 315 316 317 318 319 320 321 322 323 324 325 326 327 328 329 330 331 332 333 334 335 336 337 338 339 340 341 342 343 344 345 346 347 348 349 350 351 352 353 354 355 356 357 358 359 360 361 362 363 364 365 366 367 368 369 370 371 372 373 374 375 376 377 378 379 380 381 382 383 384 385 386 387 388 389 390 391 392 393 394 395 396 397 398 399 400 401 402 403 404 405 406 407 408 409 410 411 412 413 414 415 416 417 418 419 420 421 422 423 424 425 426 427 428 429 430 431 432 433 434 435 436 437 438 439 440 441 442 443 444 445 446 447 448 449 450 451 452 453 454 455 456 457 458 459 460 461 462 463 464 465 466 467 468 469 470 471 472 473 474 475 476 477 478 479 480 481 482 | #!/usr/bin/env python3
# License: CC-BY-NC-ND-4.0
# Created by: Patrick Lumbantobing, Vertox-AI
# Copyright (c) 2026 Vertox-AI. All rights reserved.
#
# This work is licensed under the Creative Commons
# Attribution-NonCommercial-NoDerivatives 4.0 International License.
# To view a copy of this license, visit
# http://creativecommons.org/licenses/by-nc-nd/4.0/
"""
Cache-aware streaming audio and feature buffers for Nemotron ASR.
Adapted from: https://github.com/NVIDIA-NeMo/NeMo/tree/main
Implements:
- :class:`CacheAwareStreamingAudioBuffer` for audio → feature chunks
compatible with NeMo cache-aware encoders.
- :class:`CacheAwareStreamingASR` for encoder/decoder state management,
hypothesis accumulation, and timestamped text output.
"""
from __future__ import annotations
import re
from collections.abc import Iterable
from typing import Generator, List, Optional
import numpy as np
import numpy.typing as npt
from src.asr.cache_aware_modules_config import (CacheAwareStreamingConfig,
TimestampedResult)
from src.asr.utils import log_softmax
LOG_ZERO_GUARD_VALUE = float(2**-24)
class CacheAwareStreamingAudioBuffer:
"""
Streaming audio and feature buffer for cache-aware ASR.
Handles:
- Chunking raw audio into overlapping frames for the preprocessor.
- Dropping padded STFT frames after the first chunk.
- Maintaining a feature buffer with pre-encode cache appended.
"""
def __init__(self, preprocessor, streaming_cfg: CacheAwareStreamingConfig) -> None:
"""
Parameters
----------
preprocessor :
Callable that maps ``(waveforms, lengths)`` to
``(features, feature_lengths)``.
streaming_cfg :
Cache-aware streaming configuration.
"""
self._preprocessor = preprocessor
self._streaming_cfg = streaming_cfg
self.audio_buffer: Optional[npt.NDArray[np.float32]] = None
self.audio_step: int = 0
self.features_buffer: Optional[npt.NDArray[np.float32]] = None
self._audio_chunks_lens = np.array(
[self._streaming_cfg.audio_chunk_frames * self._streaming_cfg.audio_frame_size],
dtype=np.int64,
)
self._audio_frames_drops_lens = (
self._streaming_cfg.audio_chunk_frames_drop * self._streaming_cfg.audio_frame_size
)
self._features_frames_takes_lens = self._streaming_cfg.audio_chunk_frames - 1
self._chunk_size = self._streaming_cfg.chunk_size[1]
self._shift_size = self._streaming_cfg.shift_size[1]
self._pre_encode_cache_size = self._streaming_cfg.pre_encode_cache_size[1]
self._cache_chunk_size = self._pre_encode_cache_size + self._chunk_size
self._features_chunk_lengths = np.array([self._cache_chunk_size], dtype=np.int64)
self._current_text: str = ""
self._first_cache_pre_encode = np.log(
np.zeros(
(1, self._streaming_cfg.input_features, self._pre_encode_cache_size),
dtype=np.float32,
)
+ LOG_ZERO_GUARD_VALUE
)
def len_audio_buffer(self) -> int:
"""Return current audio buffer length (samples)."""
return int(self.audio_buffer.shape[-1]) if self.audio_buffer is not None else 0
def len_features_buffer(self) -> int:
"""Return current feature buffer length (frames)."""
return int(self.features_buffer.shape[-1]) if self.features_buffer is not None else 0
def reset_buffers(self) -> None:
"""Reset both audio and feature buffers."""
self.reset_audio_buffer()
self.reset_features_buffer()
def reset_audio_buffer(self) -> None:
"""Reset audio buffer and step counter."""
self.audio_buffer = None
self.audio_step = 0
def reset_features_buffer(self) -> None:
"""Reset feature buffer."""
self.features_buffer = None
def append_audio_buffer(self, audio_signal: npt.NDArray[np.float32]) -> None:
"""Append new audio samples to the buffer."""
if self.audio_buffer is None:
self.audio_buffer = audio_signal
else:
self.audio_buffer = np.concatenate((self.audio_buffer, audio_signal), axis=-1).astype(np.float32)
def process_audio_buffer(
self,
last: bool = False,
) -> Generator[Optional[npt.NDArray[np.float32]], None, None]:
"""
Convert buffered audio into feature chunks.
Yields
------
np.ndarray or None
Feature chunks of shape ``(1, feats, frames)`` or ``None`` when
no more chunks are available.
"""
if self.audio_buffer is None:
if last:
yield None
return
while self._audio_chunks_lens[0] <= self.audio_buffer.shape[-1]:
audio_chunks = self.audio_buffer[:, : self._audio_chunks_lens[0]]
audio_features, _ = self._preprocessor(audio_chunks, self._audio_chunks_lens)
self.audio_buffer = self.audio_buffer[:, self._audio_frames_drops_lens :]
if self.audio_step > 0:
audio_features = audio_features[
:,
:,
self._streaming_cfg.audio_chunk_frames_drop : self._features_frames_takes_lens,
]
else:
audio_features = audio_features[:, :, : self._features_frames_takes_lens]
self.audio_step += self._audio_frames_drops_lens
yield audio_features
if last and self.audio_buffer is not None and self.audio_buffer.shape[-1] > 0:
n_pad = self._audio_chunks_lens[0] - self.audio_buffer.shape[-1]
zeros_pad = np.zeros((1, n_pad), dtype=np.float32)
self.audio_buffer = np.concatenate((self.audio_buffer, zeros_pad), axis=-1).astype(np.float32)
audio_chunks = self.audio_buffer[:, : self._audio_chunks_lens[0]]
audio_features, _ = self._preprocessor(audio_chunks, self._audio_chunks_lens)
self.audio_buffer = self.audio_buffer[:, self._audio_chunks_lens[0] :]
if self.audio_step > 0:
yield audio_features[:, :, self._streaming_cfg.audio_chunk_frames_drop :]
else:
yield audio_features
self.reset_audio_buffer()
yield None
def append_audio_buffer_to_process_for_features(
self,
audio_signal: npt.NDArray[np.float32],
last: bool = False,
) -> Generator[Optional[npt.NDArray[np.float32]], None, None]:
"""Append audio and immediately yield any ready feature chunks."""
self.append_audio_buffer(audio_signal)
return self.process_audio_buffer(last=last)
def append_features_buffer(self, audio_features: npt.NDArray[np.float32]) -> None:
"""Append new feature frames, preprending initial pre-encode cache if needed."""
if self.features_buffer is None:
self.features_buffer = np.concatenate((self._first_cache_pre_encode, audio_features), axis=-1).astype(
np.float32
)
else:
self.features_buffer = np.concatenate((self.features_buffer, audio_features), axis=-1).astype(np.float32)
def process_features_buffer(
self,
last: bool = False,
) -> Generator[Optional[npt.NDArray[np.float32]], None, None]:
"""
Convert feature buffer into encoder-ready feature chunks.
Yields
------
np.ndarray or None
Feature chunks of shape ``(1, feats, cache_chunk_size)`` or
``None`` when no more chunks are available.
"""
if self.features_buffer is None:
if last:
yield None
return
while self._cache_chunk_size <= self.features_buffer.shape[-1]:
features_chunk = self.features_buffer[:, :, : self._cache_chunk_size]
self.features_buffer = self.features_buffer[:, :, self._shift_size :]
yield features_chunk
if last and self.features_buffer.shape[-1] > 0:
n_pad = self._cache_chunk_size - self.features_buffer.shape[-1]
zeros_pad = np.log(
np.zeros(
(1, self.features_buffer.shape[1], n_pad),
dtype=np.float32,
)
+ LOG_ZERO_GUARD_VALUE
)
features_chunk = np.concatenate((self.features_buffer, zeros_pad), axis=-1).astype(np.float32)
self.features_buffer = self.features_buffer[:, :, self._cache_chunk_size :]
yield features_chunk
self.reset_features_buffer()
yield None
def append_features_buffer_to_process_for_features_chunk(
self,
audio_features: npt.NDArray[np.float32],
last: bool = False,
) -> Generator[Optional[npt.NDArray[np.float32]], None, None]:
"""Append features and immediately yield any ready feature chunks."""
self.append_features_buffer(audio_features)
return self.process_features_buffer(last=last)
class CacheAwareStreamingASR:
"""
Cache-aware streaming ASR wrapper around encoder/decoder ONNX models.
Maintains encoder caches, decoder recurrent state, and an evolving
hypothesis (tokens, timestamps, logprobs), producing incremental
:class:`TimestampedResult` objects from feature chunks.
"""
def __init__(
self,
asr_encoder,
asr_decoder,
vocab: List[int],
blank_idx: int,
streaming_cfg: CacheAwareStreamingConfig,
) -> None:
"""
Parameters
----------
asr_encoder :
ONNX Runtime session for the cache-aware encoder.
asr_decoder :
ONNX Runtime session for the decoder/joint network.
vocab :
Mapping from token IDs to text pieces.
blank_idx :
Index of the blank label in the vocabulary.
streaming_cfg :
Cache-aware streaming configuration.
"""
self._asr_encoder = asr_encoder
self._asr_decoder = asr_decoder
self._vocab = vocab
self._vocab_size = len(self._vocab)
self._blank_idx = blank_idx
self._streaming_cfg = streaming_cfg
# encoder cache
self._cache_last_channel: npt.NDArray[np.float32] | None = None
self._cache_last_time: npt.NDArray[np.float32] | None = None
self._cache_last_channel_len: npt.NDArray[np.int64] | None = None
self.set_init_encoder_cache()
# encoder lengths
self._chunk_size = self._streaming_cfg.chunk_size[1]
self._pre_encode_cache_size = self._streaming_cfg.pre_encode_cache_size[1]
self._cache_chunk_size = self._pre_encode_cache_size + self._chunk_size
self._features_chunk_lengths = np.array([self._cache_chunk_size], dtype=np.int64)
self._encoder_out_lengths = np.array(
[self._streaming_cfg.valid_encoder_out_len],
dtype=np.int64,
)
# decoder state
self._prev_state: tuple[npt.NDArray[np.float32], npt.NDArray[np.float32]] | None = None
self._tokens: List[int] | None = None
self._timestamps: List[int] | None = None
self._logprobs: List[float] | None = None
self._t_index: int | None = None
self.set_init_decoder_state()
self.set_init_decoder_vars()
self._current_text: str = ""
self._DECODE_SPACE_PATTERN = re.compile(r"\A\s|\s\B|(\s)\b")
def set_init_encoder_cache(self) -> None:
"""Initialise encoder caches to zeros."""
self._cache_last_channel = np.zeros(
(
self._streaming_cfg.len_layers,
1,
self._streaming_cfg.last_channel_cache_size,
self._streaming_cfg.d_model,
),
dtype=np.float32,
).transpose(1, 0, 2, 3)
self._cache_last_time = np.zeros(
(
self._streaming_cfg.len_layers,
1,
self._streaming_cfg.d_model,
self._streaming_cfg.conv_context_size[0],
),
dtype=np.float32,
).transpose(1, 0, 2, 3)
self._cache_last_channel_len = np.zeros(1, dtype=np.int64)
def set_init_decoder_state(self) -> None:
"""Initialise decoder hidden states to zeros based on input shapes."""
shapes = {x.name: x.shape for x in self._asr_decoder.get_inputs()}
self._prev_state = (
np.zeros(
shape=(shapes["input_states_1"][0], 1, shapes["input_states_1"][2]),
dtype=np.float32,
),
np.zeros(
shape=(shapes["input_states_2"][0], 1, shapes["input_states_2"][2]),
dtype=np.float32,
),
)
def set_init_decoder_vars(self) -> None:
"""Reset token, timestamp, logprob lists and time index."""
self._tokens = []
self._timestamps = []
self._logprobs = []
self._t_index = 0
def reset_states(self) -> None:
"""Reset encoder cache, decoder state, and current text."""
self.set_init_encoder_cache()
self.set_init_decoder_state()
self.set_init_decoder_vars()
self._current_text = ""
def process_encoder_step(
self,
features_chunk: npt.NDArray[np.float32],
) -> npt.NDArray[np.float32]:
"""
Run one encoder step with cache-aware inputs.
Returns
-------
encoder_out: ``(batch, time, dimension)``
"""
assert self._features_chunk_lengths[0] == features_chunk.shape[-1]
(
encoder_out,
encoder_out_lens,
cache_last_channel_next,
cache_last_time_next,
cache_last_channel_next_len,
) = self._asr_encoder.run(
[
"outputs",
"encoded_lengths",
"cache_last_channel_next",
"cache_last_time_next",
"cache_last_channel_next_len",
],
{
"audio_signal": features_chunk,
"length": self._features_chunk_lengths,
"cache_last_channel": self._cache_last_channel,
"cache_last_time": self._cache_last_time,
"cache_last_channel_len": self._cache_last_channel_len,
},
)
self._cache_last_channel = cache_last_channel_next
self._cache_last_time = cache_last_time_next
self._cache_last_channel_len = cache_last_channel_next_len
return encoder_out.transpose(0, 2, 1)
def _decode_tokens(
self, ids: Iterable[int], indices: Iterable[int] | None, logprobs: Iterable[float] | None
) -> TimestampedResult:
"""
Decode token ids including timestamps, running text, and text delta.
Returns
-------
TimestampedResult:
contains running text, timestamps, all tokens, all logprobs, and text delta
"""
tokens = [self._vocab[i] for i in ids]
text = re.sub(self._DECODE_SPACE_PATTERN, lambda x: " " if x.group(1) else "", "".join(tokens))
n_added_chars = len(text) - len(self._current_text)
added_text = text[-n_added_chars:] if n_added_chars > 0 else ""
timestamps = (
None
if indices is None
else (
self._streaming_cfg.window_step * self._streaming_cfg.subsampling_factor * np.asarray(indices)
).tolist()
)
return TimestampedResult(
text, timestamps, tokens, None if logprobs is None else np.asarray(logprobs).tolist(), added_text
)
def process_decoder_step(self, encoder_out):
"""
Run decoder steps with chunked encoder output.
Returns
-------
text: string
full transcript from the start
added_text: string
text delta
"""
encodings = encoder_out[0]
encodings_len = self._encoder_out_lengths[0]
assert encodings_len == encodings.shape[0]
step = 0
emitted_tokens = 0
while step < encodings_len:
outputs, state1, state2 = self._asr_decoder.run(
["outputs", "output_states_1", "output_states_2"],
{
"encoder_outputs": encodings[step : step + 1, :, None],
"targets": [[self._tokens[-1] if self._tokens else self._blank_idx]],
"target_length": [1],
"input_states_1": self._prev_state[0],
"input_states_2": self._prev_state[1],
},
)
logits = outputs.squeeze()
state = (state1, state2)
assert logits.shape[-1] <= self._vocab_size
token = logits.argmax()
if token != self._blank_idx:
self._prev_state = state
self._tokens.append(int(token))
self._timestamps.append(self._t_index)
emitted_tokens += 1
self._logprobs.append(log_softmax(logits)[token])
if token == self._blank_idx or emitted_tokens == self._streaming_cfg.max_tokens_per_step:
self._t_index += 1
emitted_tokens = 0
step += 1
if len(self._tokens) > 0:
res = self._decode_tokens(self._tokens, self._timestamps, self._logprobs)
self._current_text = res.text
return res.text, res.added_text
else:
return None, None
|