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Samples should be an integer, not seconds

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- Merge pull request #49 from skripnik/patch-1
- tested performance --  ESIC dev2, 27 docs, on En, De, Cs ASR, Nvidia A40, min chunk 1s, VAD => it has lower WER and latency with "segment" buffer trimming with various thresholds
This commit is contained in:
Dominik Macháček
2024-01-03 10:37:32 +01:00
committed by GitHub
parent aa51e39de4 234ac8f5e8
commit bfbe83d792
1 changed files with 1 additions and 1 deletions
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whisper_online.py
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@@ -355,7 +355,7 @@ class OnlineASRProcessor:
"""
self.transcript_buffer.pop_commited(time)
cut_seconds = time - self.buffer_time_offset
self.audio_buffer = self.audio_buffer[int(cut_seconds)*self.SAMPLING_RATE:]
self.audio_buffer = self.audio_buffer[int(cut_seconds*self.SAMPLING_RATE):]
self.buffer_time_offset = time
self.last_chunked_at = time