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import sys
import numpy as np
import logging
from typing import List, Tuple, Optional
from whisperlivekit.timed_objects import ASRToken, Sentence, Transcript
logger = logging.getLogger(__name__)
class HypothesisBuffer:
"""
Buffer to store and process ASR hypothesis tokens.
It holds:
- committed_in_buffer: tokens that have been confirmed (committed)
- buffer: the last hypothesis that is not yet committed
- new: new tokens coming from the recognizer
"""
def __init__(self, logfile=sys.stderr, confidence_validation=False):
self.confidence_validation = confidence_validation
self.committed_in_buffer: List[ASRToken] = []
self.buffer: List[ASRToken] = []
self.new: List[ASRToken] = []
self.last_committed_time = 0.0
self.last_committed_word: Optional[str] = None
self.logfile = logfile
def insert(self, new_tokens: List[ASRToken], offset: float):
"""
Insert new tokens (after applying a time offset) and compare them with the
already committed tokens. Only tokens that extend the committed hypothesis
are added.
"""
# Apply the offset to each token.
new_tokens = [token.with_offset(offset) for token in new_tokens]
# Only keep tokens that are roughly “new”
self.new = [token for token in new_tokens if token.start > self.last_committed_time - 0.1]
if self.new:
first_token = self.new[0]
if abs(first_token.start - self.last_committed_time) < 1:
if self.committed_in_buffer:
committed_len = len(self.committed_in_buffer)
new_len = len(self.new)
# Try to match 1 to 5 consecutive tokens
max_ngram = min(min(committed_len, new_len), 5)
for i in range(1, max_ngram + 1):
committed_ngram = " ".join(token.text for token in self.committed_in_buffer[-i:])
new_ngram = " ".join(token.text for token in self.new[:i])
if committed_ngram == new_ngram:
removed = []
for _ in range(i):
removed_token = self.new.pop(0)
removed.append(repr(removed_token))
logger.debug(f"Removing last {i} words: {' '.join(removed)}")
break
def flush(self) -> List[ASRToken]:
"""
Returns the committed chunk, defined as the longest common prefix
between the previous hypothesis and the new tokens.
"""
committed: List[ASRToken] = []
while self.new:
current_new = self.new[0]
if self.confidence_validation and current_new.probability and current_new.probability > 0.95:
committed.append(current_new)
self.last_committed_word = current_new.text
self.last_committed_time = current_new.end
self.new.pop(0)
self.buffer.pop(0) if self.buffer else None
elif not self.buffer:
break
elif current_new.text == self.buffer[0].text:
committed.append(current_new)
self.last_committed_word = current_new.text
self.last_committed_time = current_new.end
self.buffer.pop(0)
self.new.pop(0)
else:
break
self.buffer = self.new
self.new = []
self.committed_in_buffer.extend(committed)
return committed
def pop_committed(self, time: float):
"""
Remove tokens (from the beginning) that have ended before `time`.
"""
while self.committed_in_buffer and self.committed_in_buffer[0].end <= time:
self.committed_in_buffer.pop(0)
class OnlineASRProcessor:
"""
Processes incoming audio in a streaming fashion, calling the ASR system
periodically, and uses a hypothesis buffer to commit and trim recognized text.
The processor supports two types of buffer trimming:
- "sentence": trims at sentence boundaries (using a sentence tokenizer)
- "segment": trims at fixed segment durations.
"""
SAMPLING_RATE = 16000
def __init__(
self,
asr,
tokenize_method: Optional[callable] = None,
buffer_trimming: Tuple[str, float] = ("segment", 15),
confidence_validation = False,
logfile=sys.stderr,
):
"""
asr: An ASR system object (for example, a WhisperASR instance) that
provides a `transcribe` method, a `ts_words` method (to extract tokens),
a `segments_end_ts` method, and a separator attribute `sep`.
tokenize_method: A function that receives text and returns a list of sentence strings.
buffer_trimming: A tuple (option, seconds), where option is either "sentence" or "segment".
"""
self.asr = asr
self.tokenize = tokenize_method
self.logfile = logfile
self.confidence_validation = confidence_validation
self.init()
self.buffer_trimming_way, self.buffer_trimming_sec = buffer_trimming
if self.buffer_trimming_way not in ["sentence", "segment"]:
raise ValueError("buffer_trimming must be either 'sentence' or 'segment'")
if self.buffer_trimming_sec <= 0:
raise ValueError("buffer_trimming_sec must be positive")
elif self.buffer_trimming_sec > 30:
logger.warning(
f"buffer_trimming_sec is set to {self.buffer_trimming_sec}, which is very long. It may cause OOM."
)
def init(self, offset: Optional[float] = None):
"""Initialize or reset the processing buffers."""
self.audio_buffer = np.array([], dtype=np.float32)
self.transcript_buffer = HypothesisBuffer(logfile=self.logfile, confidence_validation=self.confidence_validation)
self.buffer_time_offset = offset if offset is not None else 0.0
self.transcript_buffer.last_committed_time = self.buffer_time_offset
self.committed: List[ASRToken] = []
def get_audio_buffer_end_time(self) -> float:
"""Returns the absolute end time of the current audio_buffer."""
return self.buffer_time_offset + (len(self.audio_buffer) / self.SAMPLING_RATE)
def insert_audio_chunk(self, audio: np.ndarray, audio_stream_end_time: Optional[float] = None):
"""Append an audio chunk (a numpy array) to the current audio buffer."""
self.audio_buffer = np.append(self.audio_buffer, audio)
def prompt(self) -> Tuple[str, str]:
"""
Returns a tuple: (prompt, context), where:
- prompt is a 200-character suffix of committed text that falls
outside the current audio buffer.
- context is the committed text within the current audio buffer.
"""
k = len(self.committed)
while k > 0 and self.committed[k - 1].end > self.buffer_time_offset:
k -= 1
prompt_tokens = self.committed[:k]
prompt_words = [token.text for token in prompt_tokens]
prompt_list = []
length_count = 0
# Use the last words until reaching 200 characters.
while prompt_words and length_count < 200:
word = prompt_words.pop(-1)
length_count += len(word) + 1
prompt_list.append(word)
non_prompt_tokens = self.committed[k:]
context_text = self.asr.sep.join(token.text for token in non_prompt_tokens)
return self.asr.sep.join(prompt_list[::-1]), context_text
def get_buffer(self):
"""
Get the unvalidated buffer in string format.
"""
return self.concatenate_tokens(self.transcript_buffer.buffer)
def process_iter(self) -> Tuple[List[ASRToken], float]:
"""
Processes the current audio buffer.
Returns a tuple: (list of committed ASRToken objects, float representing the audio processed up to time).
"""
current_audio_processed_upto = self.get_audio_buffer_end_time()
prompt_text, _ = self.prompt()
logger.debug(
f"Transcribing {len(self.audio_buffer)/self.SAMPLING_RATE:.2f} seconds from {self.buffer_time_offset:.2f}"
)
res = self.asr.transcribe(self.audio_buffer, init_prompt=prompt_text)
tokens = self.asr.ts_words(res)
self.transcript_buffer.insert(tokens, self.buffer_time_offset)
committed_tokens = self.transcript_buffer.flush()
self.committed.extend(committed_tokens)
completed = self.concatenate_tokens(committed_tokens)
logger.debug(f">>>> COMPLETE NOW: {completed.text}")
incomp = self.concatenate_tokens(self.transcript_buffer.buffer)
logger.debug(f"INCOMPLETE: {incomp.text}")
if committed_tokens and self.buffer_trimming_way == "sentence":
if len(self.audio_buffer) / self.SAMPLING_RATE > self.buffer_trimming_sec:
self.chunk_completed_sentence()
s = self.buffer_trimming_sec if self.buffer_trimming_way == "segment" else 30
if len(self.audio_buffer) / self.SAMPLING_RATE > s:
self.chunk_completed_segment(res)
logger.debug("Chunking segment")
logger.debug(
f"Length of audio buffer now: {len(self.audio_buffer)/self.SAMPLING_RATE:.2f} seconds"
)
return committed_tokens, current_audio_processed_upto
def chunk_completed_sentence(self):
"""
If the committed tokens form at least two sentences, chunk the audio
buffer at the end time of the penultimate sentence.
Also ensures chunking happens if audio buffer exceeds a time limit.
"""
buffer_duration = len(self.audio_buffer) / self.SAMPLING_RATE
if not self.committed:
if buffer_duration > self.buffer_trimming_sec:
chunk_time = self.buffer_time_offset + (buffer_duration / 2)
logger.debug(f"--- No speech detected, forced chunking at {chunk_time:.2f}")
self.chunk_at(chunk_time)
return
logger.debug("COMPLETED SENTENCE: " + " ".join(token.text for token in self.committed))
sentences = self.words_to_sentences(self.committed)
for sentence in sentences:
logger.debug(f"\tSentence: {sentence.text}")
chunk_done = False
if len(sentences) >= 2:
while len(sentences) > 2:
sentences.pop(0)
chunk_time = sentences[-2].end
logger.debug(f"--- Sentence chunked at {chunk_time:.2f}")
self.chunk_at(chunk_time)
chunk_done = True
if not chunk_done and buffer_duration > self.buffer_trimming_sec:
last_committed_time = self.committed[-1].end
logger.debug(f"--- Not enough sentences, chunking at last committed time {last_committed_time:.2f}")
self.chunk_at(last_committed_time)
def chunk_completed_segment(self, res):
"""
Chunk the audio buffer based on segment-end timestamps reported by the ASR.
Also ensures chunking happens if audio buffer exceeds a time limit.
"""
buffer_duration = len(self.audio_buffer) / self.SAMPLING_RATE
if not self.committed:
if buffer_duration > self.buffer_trimming_sec:
chunk_time = self.buffer_time_offset + (buffer_duration / 2)
logger.debug(f"--- No speech detected, forced chunking at {chunk_time:.2f}")
self.chunk_at(chunk_time)
return
logger.debug("Processing committed tokens for segmenting")
ends = self.asr.segments_end_ts(res)
last_committed_time = self.committed[-1].end
chunk_done = False
if len(ends) > 1:
logger.debug("Multiple segments available for chunking")
e = ends[-2] + self.buffer_time_offset
while len(ends) > 2 and e > last_committed_time:
ends.pop(-1)
e = ends[-2] + self.buffer_time_offset
if e <= last_committed_time:
logger.debug(f"--- Segment chunked at {e:.2f}")
self.chunk_at(e)
chunk_done = True
else:
logger.debug("--- Last segment not within committed area")
else:
logger.debug("--- Not enough segments to chunk")
if not chunk_done and buffer_duration > self.buffer_trimming_sec:
logger.debug(f"--- Buffer too large, chunking at last committed time {last_committed_time:.2f}")
self.chunk_at(last_committed_time)
logger.debug("Segment chunking complete")
def chunk_at(self, time: float):
"""
Trim both the hypothesis and audio buffer at the given time.
"""
logger.debug(f"Chunking at {time:.2f}s")
logger.debug(
f"Audio buffer length before chunking: {len(self.audio_buffer)/self.SAMPLING_RATE:.2f}s"
)
self.transcript_buffer.pop_committed(time)
cut_seconds = time - self.buffer_time_offset
self.audio_buffer = self.audio_buffer[int(cut_seconds * self.SAMPLING_RATE):]
self.buffer_time_offset = time
logger.debug(
f"Audio buffer length after chunking: {len(self.audio_buffer)/self.SAMPLING_RATE:.2f}s"
)
def words_to_sentences(self, tokens: List[ASRToken]) -> List[Sentence]:
"""
Converts a list of tokens to a list of Sentence objects using the provided
sentence tokenizer.
"""
if not tokens:
return []
full_text = " ".join(token.text for token in tokens)
if self.tokenize:
try:
sentence_texts = self.tokenize(full_text)
except Exception as e:
# Some tokenizers (e.g., MosesSentenceSplitter) expect a list input.
try:
sentence_texts = self.tokenize([full_text])
except Exception as e2:
raise ValueError("Tokenization failed") from e2
else:
sentence_texts = [full_text]
sentences: List[Sentence] = []
token_index = 0
for sent_text in sentence_texts:
sent_text = sent_text.strip()
if not sent_text:
continue
sent_tokens = []
accumulated = ""
# Accumulate tokens until roughly matching the length of the sentence text.
while token_index < len(tokens) and len(accumulated) < len(sent_text):
token = tokens[token_index]
accumulated = (accumulated + " " + token.text).strip() if accumulated else token.text
sent_tokens.append(token)
token_index += 1
if sent_tokens:
sentence = Sentence(
start=sent_tokens[0].start,
end=sent_tokens[-1].end,
text=" ".join(t.text for t in sent_tokens),
)
sentences.append(sentence)
return sentences
def finish(self) -> Tuple[List[ASRToken], float]:
"""
Flush the remaining transcript when processing ends.
Returns a tuple: (list of remaining ASRToken objects, float representing the final audio processed up to time).
"""
remaining_tokens = self.transcript_buffer.buffer
logger.debug(f"Final non-committed tokens: {remaining_tokens}")
final_processed_upto = self.buffer_time_offset + (len(self.audio_buffer) / self.SAMPLING_RATE)
self.buffer_time_offset = final_processed_upto
return remaining_tokens, final_processed_upto
def concatenate_tokens(
self,
tokens: List[ASRToken],
sep: Optional[str] = None,
offset: float = 0
) -> Transcript:
sep = sep if sep is not None else self.asr.sep
text = sep.join(token.text for token in tokens)
probability = sum(token.probability for token in tokens if token.probability) / len(tokens) if tokens else None
if tokens:
start = offset + tokens[0].start
end = offset + tokens[-1].end
else:
start = None
end = None
return Transcript(start, end, text, probability=probability)
class VACOnlineASRProcessor:
"""
Wraps an OnlineASRProcessor with a Voice Activity Controller (VAC).
It receives small chunks of audio, applies VAD (e.g. with Silero),
and when the system detects a pause in speech (or end of an utterance)
it finalizes the utterance immediately.
"""
SAMPLING_RATE = 16000
def __init__(self, online_chunk_size: float, *args, **kwargs):
self.online_chunk_size = online_chunk_size
self.online = OnlineASRProcessor(*args, **kwargs)
self.asr = self.online.asr
# Load a VAD model (e.g. Silero VAD)
import torch
model, _ = torch.hub.load(repo_or_dir="snakers4/silero-vad", model="silero_vad")
from .silero_vad_iterator import FixedVADIterator
self.vac = FixedVADIterator(model)
self.logfile = self.online.logfile
self.last_input_audio_stream_end_time: float = 0.0
self.init()
def init(self):
self.online.init()
self.vac.reset_states()
self.current_online_chunk_buffer_size = 0
self.last_input_audio_stream_end_time = self.online.buffer_time_offset
self.is_currently_final = False
self.status: Optional[str] = None # "voice" or "nonvoice"
self.audio_buffer = np.array([], dtype=np.float32)
self.buffer_offset = 0 # in frames
def get_audio_buffer_end_time(self) -> float:
"""Returns the absolute end time of the audio processed by the underlying OnlineASRProcessor."""
return self.online.get_audio_buffer_end_time()
def clear_buffer(self):
self.buffer_offset += len(self.audio_buffer)
self.audio_buffer = np.array([], dtype=np.float32)
def insert_audio_chunk(self, audio: np.ndarray, audio_stream_end_time: float):
"""
Process an incoming small audio chunk:
- run VAD on the chunk,
- decide whether to send the audio to the online ASR processor immediately,
- and/or to mark the current utterance as finished.
"""
self.last_input_audio_stream_end_time = audio_stream_end_time
res = self.vac(audio)
self.audio_buffer = np.append(self.audio_buffer, audio)
if res is not None:
# VAD returned a result; adjust the frame number
frame = list(res.values())[0] - self.buffer_offset
if "start" in res and "end" not in res:
self.status = "voice"
send_audio = self.audio_buffer[frame:]
self.online.init(offset=(frame + self.buffer_offset) / self.SAMPLING_RATE)
self.online.insert_audio_chunk(send_audio)
self.current_online_chunk_buffer_size += len(send_audio)
self.clear_buffer()
elif "end" in res and "start" not in res:
self.status = "nonvoice"
send_audio = self.audio_buffer[:frame]
self.online.insert_audio_chunk(send_audio)
self.current_online_chunk_buffer_size += len(send_audio)
self.is_currently_final = True
self.clear_buffer()
else:
beg = res["start"] - self.buffer_offset
end = res["end"] - self.buffer_offset
self.status = "nonvoice"
send_audio = self.audio_buffer[beg:end]
self.online.init(offset=(beg + self.buffer_offset) / self.SAMPLING_RATE)
self.online.insert_audio_chunk(send_audio)
self.current_online_chunk_buffer_size += len(send_audio)
self.is_currently_final = True
self.clear_buffer()
else:
if self.status == "voice":
self.online.insert_audio_chunk(self.audio_buffer)
self.current_online_chunk_buffer_size += len(self.audio_buffer)
self.clear_buffer()
else:
# Keep 1 second worth of audio in case VAD later detects voice,
# but trim to avoid unbounded memory usage.
self.buffer_offset += max(0, len(self.audio_buffer) - self.SAMPLING_RATE)
self.audio_buffer = self.audio_buffer[-self.SAMPLING_RATE:]
def process_iter(self) -> Tuple[List[ASRToken], float]:
"""
Depending on the VAD status and the amount of accumulated audio,
process the current audio chunk.
Returns a tuple: (list of committed ASRToken objects, float representing the audio processed up to time).
"""
if self.is_currently_final:
return self.finish()
elif self.current_online_chunk_buffer_size > self.SAMPLING_RATE * self.online_chunk_size:
self.current_online_chunk_buffer_size = 0
return self.online.process_iter()
else:
logger.debug("No online update, only VAD")
return [], self.last_input_audio_stream_end_time
def finish(self) -> Tuple[List[ASRToken], float]:
"""
Finish processing by flushing any remaining text.
Returns a tuple: (list of remaining ASRToken objects, float representing the final audio processed up to time).
"""
result_tokens, processed_upto = self.online.finish()
self.current_online_chunk_buffer_size = 0
self.is_currently_final = False
return result_tokens, processed_upto
def get_buffer(self):
"""
Get the unvalidated buffer in string format.
"""
return self.online.concatenate_tokens(self.online.transcript_buffer.buffer)
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