Hello Pierre, That's good info, particularly for mixing and mastering music.
The problem I have with generating stimuli for research is that is can't be "mixed for taste" -- that is, compression, reverb, and EQ shouldn't be used to make the recording sound better. In an attempt to help my fellow students who are also doing hearing research, I outlined several things that should be considered when producing "real world" stimuli. In the outline, I had stated that nearly every Grammy-winning pop recording used compression and "verb", whether hardware-based or VST/RTAS (I continue to use a Teletronix LA-2A for vocalists). But we can't use compressed speech or environmental sounds if we wish to replicate environments... unless it's the effects of compression we wish to study. Recording speech in a restaurant is trickier than one might imagine. In a noisy environment, the speech alone has a wide dynamic range. The average rms value for speech (65 dBA) and it dynamic range based on rms values is meaningless. I don't actually know what the range is when we compare the softest phoneme to the loudest voiced sound or to a raised voice. Naturally, we raise our voices above the background noise (the Lombard effect), and one paper by Tufts and Frank (2003) showed that a talker’s voice level, on average, increases 5 dB for every 10 dB increase in background noise level. Reference: Tufts, J. B., and Frank, T. (2003). Speech production in noise with and without hearing protection. J. Acoust. Soc. Am., 114(2), 1069-1080. When we combine speech with a cacophony of background noise, managing the recording without compression or clipping becomes a challenge. Naturally, compression at the first stage of amplification would help a great deal here, and may go unnoticed (perceptually) when using a curvilinear compression with low compression ratio. But if I then wish to use this as "real-world" stimuli to study the effects of hearing aid compression on (for example) localization or speech intelligibility in noise, I can't say the hearing aid is doing the work--compression had already been applied. So, modifications of any kind or psychoacoustic anomolies that aren't present in real-world scenarios taint the research stimuli. I can't use my own ears to "hear" what someone with a disorder might hear. But if I know with reasonable certainty that the stimuli PHYSICALLY reflects real-world conditions, then the problems a hearing-impaired person faces in everyday listening should be replicated in a controlled, laboratory condition. Thanks again for passing along your two cents--it really is valuable to know what goes on the "regular" world of music recording. I have Altiverb, but really haven't used it much. Now I'm interested in exploring it further. Kind regards, Eric C. ________________________________ From: Pierre Alexandre Tremblay <tremb...@gmail.com> To: Eric Carmichel <e...@elcaudio.com>; Surround Sound discussion group <sursound@music.vt.edu> Cc: Fons Adriaensen <f...@linuxaudio.org> Sent: Monday, December 17, 2012 3:20 AM Subject: Re: [Sursound] Plate Reverb rocks You know, one thing not to forget is that most pop music mixer will pre- and post-process reverb sends... filtering and compression are very common before the reverb, and would smooth the transient triggering the reverb... the opposite can also be true... I use a chorus before or after the reverb too to help the static-ness of IRs... now Altiverb has built it in... my 2 cents -------------- next part -------------- An HTML attachment was scrubbed... URL: <https://mail.music.vt.edu/mailman/private/sursound/attachments/20121217/432331d9/attachment.html> _______________________________________________ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
