I believe the buffer(mine is 26ms) does not necessarily mean the audio latency(delay). Instead, I thinks it's used for continuous DAC process for analog speaker connection or SPDIF encapsulation for digital connection.Once the buffer(FIFO) starts to be filled with the minimum pcm segments in a brief time(0ms, no sync), it will ensure the seemingless rendering process(sync) unless there's any major resource holding longer than the buffer length.Unfortunately, the FFDShow's 'dolby decoder' DSP does not seem to be DPL II yet. It looks like DPL or DS, identical rears.Also remember, any s/w dpl(II) emulations(esp. Decoding), can never be as good as the h/w solutions which also cannot be perfect in reproduction of original channels(i.e. Servo feedback).The analog line-in is converted to PCM through ADC while the SPDIF-in is de-capsulated to PCM, to be fed to the capture device.
Other than the quality(fidelity,carrier., SPDIF is pure DIGITAL), they are pretty much same in the later processes(almost no delay in encoding/decoding unlike DD). Quote:1) Yes it can.2) Why do you think that it isn't possible?3) Our algorithm is as good (please test)BTW. FFDshow has some simplified algorithm,therefore it does not work as it should(no front vs rear channels separation)Cheers, 3d.The major differences I assume are:1) 180 degree phase shift(rear merge in most s/w solutions) vs. 90 degree(h/w enc/dec)2) The servo feedback process for adjusting surrounds(center,rear) in h/w decodingThe rests seem to be what you need to prove(system model) prior to any claims or tests.Do you add any ancillary data to the DPL II stream for later better decoding?Because I read somewhere claiming it(emulation) decodes better than h/w(original, certified) decoders. It is true that most of the software solutions do not work.Also it does not mean that it is not possible to implementa proper decoder (in fact any analog or DSP h/w can be implemented as software DSP).Emulating analogue filters is a tricky business, and people claimthat digial will never match analog filters quality, but this is a different story and it does not concern DPLII that much.(it more related to emulating some special synthesizer filters, etc.)BTW. I wonder if modern amps have DSP chips for DPLII decoding or analog circuits? I think DSP.No, there are no 'dirty' tricks in our decoder(no auxilary data).The quality is good because it operates in bark scale (critical filterbanks), so it is a 'improved' sorta solution.
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The DPLII concept is pretty old, and I have used some modern DSP solutions (bark scale approach, etc), which are not present in certified dolby algorithm.To prove that the algorithm sounds better than certified dolby algorithm would require descent subjective tests.So someone would have to do such an evaluation.However users can simply listen and decide how does it compare to for example certified PowerDVD algorithm. And the difference is significant!!(I hope you will have a chance to check it out)Cheers, 3d.

@Rocaria.Well, in fact I have no idea how the original DPLII decoder works.I knew how to downmix (thanx to your post, by the way ),and I have some DSP knowledge and experience, so figured outhow to resolve that. Since I was thinking in terms of the latestDSP techniques, possibly my approach is a bit more modern that Dolby's approach. And this may result in better quality-@Radosuaf and Voidming.I am sorry, I am not a DS expert. Nor soundcard expert.I have heard that there are problems with 7.1 soundcards.But maybe someone else can help.Thanks. So if your DSFilter is not emulating the DPL II decoding, it's better be called as (DPL II) upmix or similar, but not 'dpl II decoding'. Just my thoughts.Yes, exactly.But I did not want to make things too complicated.And since it is a sort of 'reverse' process - upmix (to the DPLII downmix), I called it DPLII decoding.I hope you can forgive me that-BTW.

Since I know that you are a user of the HRTF based emulation implemented in FFDshow,I am curious of your thoughs regarding my virtual surround emulatioh.Please check out this thread.
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