Waves De Emphasis 16bit
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The de-emphasis buffer size should be set by the user. Since it's just multiplying the audio data by a factor, it's easy to do in software. I found that using deq (http://www.jwz.org/deq/) to get a decent de-emphasis setting is a good way to go. The deq program searches the subcode and the TOC for pre-emphasis flags (if any) and applies a de-emphasis factor to the audio data as specified in the flags. If deq does not find any flags, it just uses a fixed value of 0.75. The deq program is available from http://www.jwz.org/deq/.
The speed of the de-emphasis process is controlled by the de-emphasis buffer size. This is usually the biggest thing I hear people complain about. Since the amount of de-emphasis is relatively small, it can easily be done in a single pass through the audio data. However, a lot of people seem to want the de-emphasis to be done more than once, most often because they want to have different de-emphasis settings per track. I think this is a bit unrealistic, because it means you'd have to scan through the audio data multiple times (or at least one time per track, plus maybe twice for the first and last few tracks). A more practical approach is probably to multiply the audio data once, while making sure the buffer is big enough for the whole set of tracks. If you have a buffer that's too small to hold all the tracks, then some tracks will be de-emphasized too much, and others too little. However, this will at least be more predictable than multiplying the data multiple times.
De-emphasis is pretty easy to do: we can simply multiply the audio data by something between 0 and 1, usually something like 0.75 (to multiply the 50% point of the bit-depth curve by 75%). The de-emphasis factor has to be adjusted to your equipment, usually based on the number of tracks, sample rate, bits per sample, bit depth and so on.
I'm not entirely sure how pre-emphasis works, but a rough estimate is that it is a fixed, low pass filter with a frequency cutoff just below the Nyquist frequency, so that the waveform starts to roll off at about half that frequency (well below the 16kHz range). A rolloff at half the Nyquist frequency suggests an effective rolloff at 4-8kHz, depending on the transients in the signal. Most of the time, digital signals will roll off at the Nyquist frequency, so I'm guessing the filter rolls off at a similar frequency.
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