Dolby Pro Logic II encoding test (90° vs 180° phase shift)

[ursamtl]

Originally Posted by 3dsnar

Yes, you've got the point. It is not so easy to implement hilbert transform.
In practice, the hilbert transform is applied via spectrum operations, not FIR filtering apprach, because convolutive filtering is a bit computationally heavy.

That's not what Dolby says about its approach. I think they are clear in their explanation of why and how for the Hilbert. They designed DPLII, etc., so I hope they know what they're talking about

(from Dolby Digital Professional Encoding Guidelines, Appendix D, page 1)

Purpose
The 90-Degree Phase Shift filter provides a means for an encoding engineer to create a multichannel Dolby Digital bitstream that can be downmixed to a Dolby Surroundcompatible Lt/Rt output. Without this filter, point-source elements panned from Surround to Center in the multichannel mix would seem to pan from Surround to Left and then to Center when downmixed to Lt/Rt and reproduced using a Dolby Surround Pro Logic decoder.

This filter should generally be used whenever encoding a multichannel signal unless it is known that the 5.1-channel source does not contain point-source element pans. For example, if the source was recorded using five discrete microphones placed in the corners of an auditorium, there is no panning between channels and the filter could be safely disabled. If in doubt, use a DP562 to downmix the 5.1-channel program to Lt/Rt, Dolby Surround Pro Logic decode the Lt/Rt signals, and then set the filter to the setting that sounds best.

Description
The 90-degree phase-shift is created using a very long FIR filter. Since this filter introduces a significant time delay, the other four channels are delayed using a PCM delay line so that all six channels are kept in sample alignment. This filter has exactly
90-degree phase shift at all frequencies. The magnitude response is flat across most of the spectrum, rolling off at the lower edge of the audio band (-3 dB below 30 Hz).

I don't know how much clearer we could get than this! Dolby clearly states how they do it: a very long FIR filter. The source code for such a filter is available around the net, either at the audiodsp.org link I provided yesterday or at a couple of other filter designer sites. Dolby also mentions that this introduces a time delay (also known as latency), so the channels not passed through the Hilbert need to be delayed to compensate for this. Maybe instead of investing all this time and energy in trying to prove one side of the argument or the other, it'd make more sense to try coding it the way Dolby describes it and see how the results turned out? Just a suggestion.

Regards,
Steve.