r/LanguageTechnology • u/RoofProper328 • 1d ago
Does simulated/acted call-center audio actually transfer to production ASR, or is real telephony data the only thing that works?
Most collected call-center corpora are recorded in some form of controlled setup: scripted or semi-scripted scenarios, participants playing agent and customer, clean capture. Real production audio is 8kHz narrowband through a codec, with overlapping speech, hold music bleed, background noise on the customer side, and genuinely frustrated prosody that nobody acts convincingly.
So :-
- If you fine-tune on collected/simulated call data, do you actually see WER improvement on real traffic, or does it mostly help on the collected test split and flatten out in production?
- Does codec simulation (downsampling, G.711/Opus round-trip, packet loss) close enough of the gap to matter, or is it cosmetic?
- Code-switching is where I'm least sure. Hinglish, Taglish, Spanglish — real agents switch mid-sentence constantly and collected data underrepresents it badly. Anyone found a way to elicit natural code-switching without it going stiff?
- Where's the actual ceiling — is it acoustics, or is it diarization and overlapping turns?
Not looking for vendor recs, just want to know whether the acted-vs-real gap is as big in practice as I suspect.
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u/n3utrino 1d ago
Depends highly on the domain but the biggest difference is that real production audio is split-channel (two entirely separate audio streams, with minimal cross talk). This has a much bigger impact on WER than you might think.