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enhance chunking to handle audio buffer time limits

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This commit is contained in:
Quentin Fuxa 2025-04-09 11:34:59 +02:00
parent 704170ccf3
commit 0b43035701
1 changed files with 40 additions and 10 deletions
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50
whisperlivekit/whisper_streaming_custom/online_asr.py
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@ -216,31 +216,54 @@ class OnlineASRProcessor:
"""
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}")
if len(sentences) < 2:
return
# Keep the last two sentences.
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 = 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
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)
@ -248,11 +271,18 @@ class OnlineASRProcessor:
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.